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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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into 13
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iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is
supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5,
only iLBC 30 is negotiated now.
ASTERISK-25309 #close
Change-Id: I92d724600a183eec3114da0ac607b994b1a793da
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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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Some codecs that may be a third party library to Asterisk need to have
knowledge of the format attributes that were negotiated. Unfortunately,
when the great format migration of Asterisk 13 occurred, that ability
was lost.
This patch adds an API call, ast_format_attribute_get, to the core
format API, along with updates to the unit test to check the new API
call. A new callback is also now available for format attribute modules,
such that they can provide the format attribute values they manage.
Note that the API returns a void *. This is done as the format attribute
modules themselves may store format attributes in any particular manner
they like. Care should be taken by consumers of the API to check the
return value before casting and dereferencing. Consumers will obviously
need to have a priori knowledge of the type of the format attribute as
well.
Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3
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use RTP." into 13
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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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* changes:
rtp_engine.c: Must protect mime_types_len with mime_types_lock.
res_pjsip_sdp_rtp.c: Fixup some whitespace.
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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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into 13
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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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This will add ECDH support to Asterisk. It will
detect auto ECDH support in OpenSSL
(1.0.2b and above) during ./configure. If this is
available, it will use it,
otherwise it will fall back to prime256v1 (this
behavior is consistent with
other projects such as Apache and nginx).
This fixes WebRTC being broken in Firefox 38+ due
to Firefox now only supporting
ciphers with perfect forward secrecy.
ASTERISK-25265 #close
Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b
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* changes:
rtp_engine.h: No sense allowing payload types larger than RFC allows.
rtp_engine.c: Minor tweaks.
rtp_engine.h: Misc comment fixes.
chan_sip.c: Tweak glue->update_peer() parameter nil value.
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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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ast_rtp_codecs_get_payload() gets called once or twice for every received
RTP frame so it would be nice to not allocate an ao2 object to then have
it destroyed shortly thereafter. The ao2 object gets allocated only if
the payload type is not set by the channel driver as a negotiated value.
The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323.
* Made static_RTP_PT[] an array of ao2 objects that
ast_rtp_codecs_get_payload() can return instead of an array of structs
that must be copied into a created ao2 object.
ASTERISK-25296 #close
Reported by: Richard Mudgett
Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0
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ASTERISK-25296
Reported by: Richard Mudgett
Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b
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Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e
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Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2
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Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973
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* Tweaked add_static_payload() to not use magic numbers.
Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b
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* Fix off nominial ref leak of new_type in
ast_rtp_codecs_payloads_set_m_type().
* No need to lock static_RTP_PT_lock in
ast_rtp_codecs_payloads_set_m_type() and
ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type
parameter sanity check.
* No need to create ast_rtp_payload_type ao2 objects with a lock since the
lock is not used.
Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4
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Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43
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Change glue->update_peer() parameter from 0 to NULL to better indicate it
is a pointer.
Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd
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Two testsuite tests crashed in the same place as a result of an INVITE
being CANCELed.
tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified
tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp
The session pointer is no longer in the inv->mod_data[session_module.id]
location because the INVITE transaction has reached the terminated state.
ASTERISK-25297 #close
Reported by: Richard Mudgett
Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427
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A test agent was continuously failing all ARI tests when run against
Asterisk 13. As it turns out, the reason for this is that on those test
runs, for some reason we decided to use the super extended 64 bit
payload length for websocket text frames instead of the extended 16 bit
payload length. For 64-bit payloads, the expected byte order over the
network is
7, 6, 5, 4, 3, 2, 1, 0
However, we were sending the payload as
3, 2, 1, 0, 7, 6, 5, 4
This meant that we were saying to expect an absolutely MASSIVE payload
to arrive. Since we did not follow through on this expected payload
size, the client would sit patiently waiting for the rest of the payload
to arrive until the test would time out.
With this change, we use the htobe64() function instead of htonl() so
that a 64-bit byte-swap is performed instead of a 32 bit byte-swap.
Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a
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This event is necessary for the bridge_wait_e_options test to be able to
confirm that ringing is being played on the local channel that runs the
BridgeWait() application with the e(r) option.
ASTERISK-25292 #close
Reported by Kevin Harwell
Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e
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Currently, if a blank musiconhold.conf is used, musiconhold will fail
to start for a channel going into a holding bridge with an anticipation
of getting music on hold. That being the case, no frames will be written
to the channel and that can pose a problem for blind transfers in PJSIP
which may rely on frames being written to get past the REFER framehook.
This patch makes holding bridges start a silence generator if starting
music on hold fails and makes it so that if no music on hold functions
are installed that the ast_moh_start function will report a failure so
that consumers of that function will be able to respond appropriately.
ASTERISK-25271 #close
Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
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Due to backwards compatible changes, the ARI version should be bumped to
1.8.0 prior to the release of 13.5.0. Note that a previous patch already
bumped the version of AMI for this release.
Change-Id: I419033bfbbc0d3533a29ccb32b2981f39e0883e7
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
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Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a
specific scenario involving local channels and a native local RTP bridge
could result in ringback still being heard on a calling channel even
after the call is bridged.
That commit caused many tests in the testsuite to fail with alarming
consequences, such as not sending DialBegin and DialEnd events, and
giving incorrect hangup causes during calls.
This commit reverts the previous commit and implements and alternate
solution. This new solution involves only passing AST_CONTROL_RINGING
frames across local channels if the local channel is in AST_STATE_RING.
Otherwise, the frame does not traverse the local channels. By doing
this, we can ensure that a playtones generator does not get started on
the calling channel but rather is started on the local channel on which
the ringing frame was initially indicated.
ASTERISK-25250 #close
Reported by Etienne Lessard
Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39
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