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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
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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
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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
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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
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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
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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
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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
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referenced"
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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
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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
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ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: I97ecebc1ab9b5654fb918bf1f4c98c956b852369
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There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk. It uses only one mapping structure to
associate payload types to codecs. The single mapping is overkill if all
of the payload type values are well known values. Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive. Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.
1) An independent payload type mapping is needed for sending and
receiving.
2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.
3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.
* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.
* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.
* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created. All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.
* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP. We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
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This is a type mismatch fix of the debugging commit
c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why
a testsuite test was failing only on one of the continuous
integration build agents.
Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75
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ASTERISK-25308 #close
Reported by: Joshua Colp
Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082
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Change-Id: I228df6adecd4cb450d03e09e9a38c86bb566e811
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* Remove extraneous unlock on off-nominal path.
* Add missing HTTP error reply.
Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b
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Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf
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When sending an RTP keepalive, we need to be sure we're not dealing with
a NULL RTP instance. There had been a NULL check, but the commit that
added the rtp_timeout and rtp_hold_timeout options removed the NULL
check.
Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64
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Due to the use of ast_websocket_close in session termination it is
possible for the underlying socket to already be closed when the
session is terminated. This occurs when the close frame is attempted
to be written out but fails.
Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b
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The res_http_websocket module will currently attempt to close
the WebSocket connection if fatal cases occur, such as when
attempting to write out data and being unable to. When the
fatal cases occur the code attempts to write a WebSocket close
frame out to have the remote side close the connection. If
writing this fails then the connection is not terminated.
This change forcefully terminates the connection if the
WebSocket is to be closed but is unable to send the close frame.
ASTERISK-25312 #close
Change-Id: I10973086671cc192a76424060d9ec8e688602845
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This patch adds the .get callback to the format attribute module, such
that the Asterisk core or other third party modules can query for the
negotiated format attributes.
Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c
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If the saved SUBSCRIBE message is not parseable for whatever reason then
Asterisk could crash when libpjsip tries to parse the message and adds an
error message to the parse error list.
* Made ast_sip_create_rdata() initialize the parse error rdata list. The
list is checked after parsing to see that it remains empty for the
function to return successful.
ASTERISK-25306
Reported by Mark Michelson
Change-Id: Ie0677f69f707503b1a37df18723bd59418085256
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We don't have a compatability function to fill in a missing htobe64; but
we already have one for the identical htonll.
Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac
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An http request can be sent to get the existing Asterisk logs.
The command "curl -v -u user:pass -X GET 'http://localhost:8088
/ari/asterisk/logging'" can be run in the terminal to access the
newly implemented functionality.
* Retrieve all existing log channels
ASTERISK-25252
Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808
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An http request can be sent to create a log channel
in Asterisk.
The command "curl -v -u user:pass -X POST
'http://localhost:088/ari/asterisk/logging/mylog?
configuration=notice,warning'" can be run in the terminal
to access the newly implemented functionality for ARI.
* Ability to create log channels using ARI
ASTERISK-25252
Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782
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An http request can be sent to delete a log channel
in Asterisk.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/logging/mylog'" can be run in the terminal
to access the newly implemented functionally for ARI.
* Able to delete log channels using ARI
ASTERISK-25252
Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
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The pjsip_rx_data structure has a pkt_info.packet field on it that is
the packet that was read from the transport. For datagram transports,
the packet read from the transport will correspond to the SIP message
that arrived. For streamed transports, however, it is possible to read
multiple SIP messages in one packet.
In a recent case, Asterisk crashed on a system where TCP was being used.
This is because at some point, a read from the TCP socket resulted in a
200 OK response as well as an incoming SUBSCRIBE request being stored in
rdata->pkt_info.packet. When the SUBSCRIBE was processed, the
combination 200 OK and SUBSCRIBE was saved in persistent storage. Later,
a restart of Asterisk resulted in the crash because the persistent
subscription recreation code ended up building the 200 OK response
instead of a SUBSCRIBE request, and we attempted to access
request-specific data.
The fix here is to use the pjsip_msg_print() function in order to
persist SUBSCRIBE requests. This way, rather than using the raw socket
data, we use the parsed SIP message that PJSIP has given us. If we
receive multiple SIP messages from a single read, we will be sure only
to save off the relevant SIP message. There also is a safeguard put in
place to make sure that if we do end up reconstructing a SIP response,
it will not cause a crash.
ASTERISK-25306 #close
Reported by Mark Michelson
Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2
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The ast_sip_sanitize_xml function is used to sanitize
a string for placement into XML. This is done by examining
an input string and then appending values to an output
buffer. The function used by its implementation, strncat,
has specific behavior that was not taken into account.
If the size of the input string exceeded the available
output buffer size it was possible for the sanitization
function to write past the output buffer itself causing
a crash. The crash would either occur because it was
writing into memory it shouldn't be or because the resulting
string was not NULL terminated.
This change keeps count of how much remaining space is
available in the output buffer for text and only allows
strncat to use that amount.
Since this was exposed by the res_pjsip_pidf_digium_body_supplement
module attempting to send a large message the maximum allowed
message size has also been increased in it.
A unit test has also been added which confirms that the
ast_sip_sanitize_xml function is providing NULL terminated
output even when the input length exceeds the output
buffer size.
ASTERISK-25304 #close
Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302
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A change recently went in which enabled perfect forward secrecy for
DTLS in res_rtp_asterisk. This was accomplished two different ways
depending on the availability of a feature in OpenSSL. The fallback
method created a temporary instance of a key but did not free it.
This change fixes that.
ASTERISK-25265
Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396
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Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a
test failure observed on 32 bit test agents by ensuring that a cast from
a 32 bit unsigned integer to a 64 bit unsigned integer was happening in
a predictable place. As it turns out, this did not cause test runs to
succeed.
This commit adds several redundant debug messages that print the payload
lengths of websocket frames. The idea here is that this commit will not
cause tests to succeed for the faulty test agent, but we might deduce
where the fault lies more easily this way by observing at what point the
expected value (537) changes to some ungangly huge number.
If you are wondering why something like this is being committed to the
branch, keep in mind that in commit
39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test
failures only happen when automated tests are run. Attempts to run the
tests by hand manually on the test agent result in the tests passing.
Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d
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We have seen a rash of test failures on a 32-bit build agent. Commit
48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where
we were not encoding a 64-bit value correctly over the wire. This
commit, however, did not solve the test failures.
In the failing tests, ARI is attempting to send a 537 byte text frame
over a websocket. When sending a frame this small, 16 bits are all that
is required in order to encode the payload length on the websocket
frame. However, ast_websocket_write() thinks that the payload length is
greater than 65535 and therefore writes out a 64 bit payload length.
Inspecting this payload length, the lower 32 bits are exactly what we
would expect it to be, 537 in hex. The upper 32 bits, are junk values
that are not expected to be there.
In the failure, we are passing the result of strlen() to a function that
expects a uint64_t parameter to be passed in. strlen() returns a size_t,
which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit
unsigned value to somewhere where a 64-bit unsigned value is expected
would cause no problems. In fact, in manual runs of failing tests, this
works just fine. However, ast_websocket_write() uses the Asterisk
optional API, which means that rather than a simple function call, there
are a series of macros that are used for its declaration and
implementation. These macros may be causing some sort of error to occur
when converting from a 32 bit quantity to a 64 bit quantity.
This commit changes the logic by making existing ast_websocket_write()
calls use ast_websocket_write_string() instead. Within
ast_websocket_write_string(), the 64-bit converted strlen is saved in a
local variable, and that variable is passed to ast_websocket_write()
instead.
Note that this commit message is full of speculation rather than
certainty. This is because the observed test failures, while always
present in automated test runs, never occur when tests are manually
attempted on the same test agent. The idea behind this commit is to fix
a theoretical issue by performing changes that should, at the least,
cause no harm. If it turns out that this change does not fix the failing
tests, then this commit should be reverted.
Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67
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An http request can be sent to rotate a specified log channel.
If the channel does not exist, an error response will be
returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/logging/logChannelName/rotate'" can be run in the
terminal to access this new functionality.
* Added the ability to rotate log files through ARI
ASTERISK-25252
Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
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Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis
applications were registered. This was done such that the WebSocket would be
ready when an application is registered. However, by creating the WebSocket
first, the client had the ability to make requests for the Stasis application
it thought had been created with the initial handshake request. The inevitable
conclusion of this scenario was the cart being put before the horse.
ASTERISK-24988 resolved half of the problem by ensuring that the applications
were created and registered with Stasis prior to completing the handshake
with the client. While this meant that Stasis was ready when the client
received the green-light from Asterisk, it also meant that the WebSocket was
not yet ready for Stasis to dispatch messages.
This patch introduces a message queuing mechanism for delaying messages from
Stasis applications while the WebSocket is being constructed. When the ARI
event processor receives the message from the WebSocket that it is being
created, the event processor instantiates an event session which contains a
message queue. It then tries to create and register the requested applications
with Stasis. Messages that are dispatched from Stasis between this point and
the point at which the event processor is notified the WebSocket is ready, are
stashed in the queue. Once the WebSocket has been built, the queue's messages
are dispatched in the order in which they were originally received and the
queue is concurrently cleared.
ASTERISK-25181 #close
Reported By: Matt Jordan
Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17
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