There's a race condition with bridging where a bridge can be torn down
causing the bridge_channel's ast_channel to become NULL when it's still
needed. This particular case happened with attended transfers, but the
crash occurred when trying to publish a stasis message. Now, the
bridge_channel is locked, a ref to the ast_channel is obtained, and that
ref is passed down the chain.
Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814
T.140 data in RTP is not zero terminated, so when we are queuing a text
frame on a bridge we need to ensure that we are passing a zero
terminated string.
ASTERISK-28974 #close
Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3
The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
at the same level as the scope level. This allows the same
messages to be printed to the debug log when AST_DEVMODE
isn't enabled.
Also added a few variants of the SCOPE_EXIT macros that will
also call ast_log instead of ast_debug to make it easier to
use scope tracing and still print error messages.
Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21
* Added ast_stream_to_stra and ast_stream_topology_to_stra() macros
which are shortcuts for
ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP))
* Added the stream position to the string representation of the
stream.
* Fixed some formatting in ast_stream_to_str().
Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b
Allow passing a topology from the called channel back to the
calling channel.
* Added a new function ast_queue_answer() that accepts a stream
topology and queues an ANSWER CONTROL frame with it as the
data. This allows the called channel to indicate its resolved
topology.
* Added a new virtual function to the channel tech structure
answer_with_stream_topology() that allows the calling channel
to receive the called channel's topology. Added
ast_raw_answer_with_stream_topology() that invokes that virtual
function.
* Modified app_dial.c and features.c to grab the topology from the
ANSWER frame queued by the answering channel and send it to
the calling channel with ast_raw_answer_with_stream_topology().
* Modified frame.c to automatically cleanup the reference
to the topology on ANSWER frames.
Added a few debugging messages to stream.c.
Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c
The assumed behavior of realloc() - that it was effectively a free() if
its second argument was 0 - is Linux specific behavior and is not
guaranteed by either POSIX or the C specification.
Instead, if we want to resize a vector to 0, do it explicitly.
Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f
With the addition of STIR/SHAKEN, the function ast_base64decode_string
was added for convenience since there is a lot of converting done during
the STIR/SHAKEN process. This function returned the decoded string for
you, but did not NULL terminate it, causing some issues (specifically
with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the
documentation has been updated to reflect this.
Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5
This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.
Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d
There are various places in Asterisk - specifically in regards to
database integration - where having some kind of UTF-8 validation would
be beneficial. This patch adds:
* Functions to validate that a given string contains only valid UTF-8
sequences.
* A function to copy a string (similar to ast_copy_string) stopping when
an invalid UTF-8 sequence is encountered.
* A UTF-8 validator that allows for progressive validation.
All of this is based on the excellent UTF-8 decoder by Björn Höhrmann.
More information is available here:
https://bjoern.hoehrmann.de/utf-8/decoder/dfa/
The API was written in such a way that should allow us to replace the
implementation later should we determine that we need something more
comprehensive.
Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9
Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.
Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.
Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.
Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.
No functional changes were made with this commit.
Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
This commit adds the endpoint options required to control
Advanced Codec Negotiation.
incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs
The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs. That'll be handled
when things finalize.
This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.
Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11
* ast_frame_subclass2str() and ast_frame_type2str() now return
a pointer to the buffer that was passed in instead of void.
This makes it easier to use these functions inline in
printf-style debugging statements.
* Added many missing control frame entries in
ast_frame_subclass2str.
Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc
Tracing through synchronous tasks was a little troublesome because
the new thread's stack counter reset to 0. This change allows
a synchronous task to set its trace level to be the same as the
thread that pushed the task. For now, the task's level has to be
passed in the task's data structure but a future enhancement to the
taskprocessor subsystem could automatically set the trace level
of the servant to be that of the caller.
This doesn't really make sense for async tasks because you never
know when they're going to run anyway.
Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825
This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.
Note, the AMI version has been bumped for this change.
ASTERISK-28945 #close
Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb
The Streams API becomes the home for the core ACN capabilities.
These include...
* Parsing and formatting of codec negotation preferences.
* Resolving pending streams and topologies with those configured
using configured preferences.
* Utility functions for creating string representations of
streams, topologies, and negotiation preferences.
For codec negotiation preferences:
* Added ast_stream_codec_prefs_parse() which takes a string
representation of codec negotiation preferences, which
may come from a pjsip endpoint for example, and populates
a ast_stream_codec_negotiation_prefs structure.
* Added ast_stream_codec_prefs_to_str() which does the reverse.
* Added many functions to parse individual parameter name
and value strings to their respectrive enum values, and the
reverse.
For streams:
* Added ast_stream_create_resolved() which takes a "live" stream
and resolves it with a configured stream and the negotiation
preferences to create a new stream.
* Added ast_stream_to_str() which create a string representation
of a stream suitable for debug or display purposes.
For topology:
* Added ast_stream_topology_create_resolved() which takes a "live"
topology and resolves it, stream by stream, with a configured
topology stream and the negotiation preferences to create a new
topology.
* Added ast_stream_topology_to_str() which create a string
representation of a topology suitable for debug or display
purposes.
* Renamed ast_format_caps_from_topology() to
ast_stream_topology_get_formats() to be more consistent with
the existing ast_stream_get_formats().
Additional changes:
* A new function ast_format_cap_append_names() appends the results
to the ast_str buffer instead of replacing buffer contents.
Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56
Created new SCOPE_ functions that don't depend on RAII_VAR. Besides
generating less code, the use of the explicit SCOPE_EXIT macros
capture the line number where the scope exited. The RAII_VAR
versions can't do that.
* SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use
RAII_VAR and therefore needs needs one of...
* SCOPE_EXIT(...): Decrements the trace stack counter and optionally
prints a message.
* SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter,
optionally prints a message, then executes the expression.
SCOPE_EXIT_EXPR(break, "My while got broken\n");
* SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter,
optionally prints a message, then returns without a value.
SCOPE_EXIT_RTN("Bye\n");
* SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace
stack counter, optionally prints a message, then returns the value
specified.
SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc);
Create an ast_str helper ast_str_tmp() that allocates a temporary
ast_str that can be passed to a function that needs it, then frees
it. This makes using the above macros easier. Example:
SCOPE_ENTER(1, Format Caps 1: %s Format Caps 2: %s\n",
ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP),
ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP));
The calls to ast_str_tmp create an ast_str of the specified initial
length which can be referenced as STR_TMP. It then calls the
expression, which must return a char *, ast_strdupa's it, frees
STR_TMP, then returns the ast_strdupa'd string. That string is
freed when the function returns.
Change-Id: I44059b20d55a889aa91440d2f8a590865998be51
Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is
sent, the caller ID will be checked to see if there is a certificate
that corresponds to it. If so, that information will be retrieved and an
Identity header will be added to the SIP message. The format is:
header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken
Header, payload, and signature are all BASE64 encoded. The public key
URL is retrieved from the certificate. Currently the algorithm and ppt
are ES256 and shaken, respectively. This message is signed and can be
used for verification on the receiving end.
Two new configuration options have been added to the certificate object:
attestation and origid. The attestation is required and must be A, B, or
C. origid is the origination identifier.
A new utility function has been added as well that takes a string,
allocates space, BASE64 encodes it, then returns it, eliminating the
need to calculate the size yourself.
Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4
This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..
This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.
This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.
Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.
ASTERISK-28899 #close
Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81
When requesting a Local channel the requested stream topology
or a converted stream topology will now be placed onto the
resulting channels.
Frames written in on streams will now also preserve the stream
identifier as they are queued on the opposite channel.
Finally when a stream topology change is requested it is
immediately accepted and reflected on both channels. Each
channel also receives a queued frame to indicate that the
topology has changed.
ASTERISK-28938
Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186
This patch fixes a few compile warnings/errors that now occur when using gcc
10+.
Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.
Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9
Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an
INVITE, the Identity header is retrieved, parsing the message to verify
the signature. If any of the parsing fails,
AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this
caller ID. If verification itself fails,
AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in
the payload does not line up with the SIP signaling,
AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps
pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the
verification process.
A new config option has been added to the general section for
stir_shaken.conf. "signature_timeout" is the amount of time a signature
will be considered valid. If an INVITE is received and the amount of
time between when it was received and when it was signed is greater than
signature_timeout, verification will fail.
Some changes were also made to signing and verification. There was an
error where the whole JSON string was being signed rather than the
header combined with the payload. This has been changed to sign the
correct thing. Verification has been changed to do this as well, and the
unit tests have been updated to reflect these changes.
A couple of utility functions have also been added. One decodes a BASE64
string and returns the decoded string, doing all the length calculations
for you. The other retrieves a string value from a header in a rdata
object.
Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913
What's wrong with ast_debug?
ast_debug is fine for general purpose debug output but it's not
really geared for scope tracing since it doesn't present its
output in a way that makes capturing and analyzing flow through
Asterisk easy.
How is scope tracing better?
Scope tracing uses the same "cleanup" attribute that RAII_VAR
uses to print messages to a separate "trace" log level. Even
better, the messages are indented and unindented based on a
thread-local call depth counter. When output to a separate log
file, the output is uncluttered and easy to follow.
Here's an example of the output. The leading timestamps and
thread ids are removed and the output cut off at 68 columns for
commit message restrictions but you get the idea.
--> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
--> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
--> res_pjsip_session.c:3669 handle_incoming_response PJSIP/
--> chan_pjsip.c:3265 chan_pjsip_incoming_response_after
--> chan_pjsip.c:3194 chan_pjsip_incoming_response P
chan_pjsip.c:3245 chan_pjsip_incoming_respon
<-- chan_pjsip.c:3194 chan_pjsip_incoming_response P
<-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after
<-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/
<-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
<-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
The messages with the "-->" or "<--" were produced by including
the following at the top of each function:
SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session));
Scope isn't limited to functions any more than RAII_VAR is. You
can also see entry and exit from "if", "for", "while", etc blocks.
There is also an ast_trace() macro that doesn't track entry or
exit but simply outputs a message to the trace log using the
current indent level. The deepest message in the sample
(chan_pjsip.c:3245) was used to indicate which "case" in a
"select" was executed.
How do you use it?
More documentation is available in logger.h but here's an overview:
* Configure with --enable-dev-mode. Like debug, scope tracing
is #ifdef'd out if devmode isn't enabled.
* Add a SCOPE_TRACE() call to the top of your function.
* Set a logger channel in logger.conf to output the "trace" level.
* Use the CLI (or cli.conf) to set a trace level similar to setting
debug level... CLI> core set trace 2 res_pjsip.so
Summary Of Changes:
* Added LOG_TRACE logger level. Actually it occupies the slot
formerly occupied by the now defunct "event" level.
* Added core asterisk option "trace" similar to debug. Includes
ability to specify global trace level in asterisk.conf and CLI
commands to turn on/off and set levels. Levels can be set
globally (probably not a good idea), or by module/source file.
* Updated sample asterisk.conf and logger.conf. Tracing is
disabled by default in both.
* Added __ast_trace() to logger.c which keeps track of the indent
level using TLS. It's #ifdef'd out if devmode isn't enabled.
* Added ast_trace() and SCOPE_TRACE() macros to logger.h.
These are all #ifdef'd out if devmode isn't enabled.
Why not use gcc's -finstrument-functions capability?
gcc's facility doesn't allow access to local data and doesn't
operate on non-function scopes.
Known Issues:
The only know issue is that we currently don't know the line
number where the scope exited. It's reported as the same place
the scope was entered. There's probably a way to get around it
but it might involve looking at the stack and doing an 'addr2line'
to get the line number. Kind of like ast_backtrace() does.
Not sure if it's worth it.
Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027
Adds the "STIR_SHAKEN" dialplan function and an API call to add a
STIR_SHAKEN verification result to a channel. This information will be
held in a datastore on the channel that can later be queried through the
"STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results
including identity, attestation, and verify_result. Here are some
examples:
STIR_SHAKEN(count)
STIR_SHAKEN(0, identity)
STIR_SHAKEN(1, attestation)
STIR_SHAKEN(2, verify_result)
Getting the count can be used to iterate through the results and pull
information by specifying the index and the field you want to retrieve.
Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82
Add a new "masquarade" channel event, and use it in app_queue to track unique id's.
Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210
ASTERISK-28829 #close
ASTERISK-25844 #close
Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
There are a lot of moving parts in this patch, but the focus of it is on
the verification of the signature using a public key located at the
public key URL provided in the JSON payload. First, we check the
database to see if we have already downloaded the key. If so, check to
see if it has expired. If it has, redownload from the URL. If we don't
have an entry in the database, just go ahead and download the public
key. The expiration is tested each time we download the file. After
that, read the public key from the file and use it to verify the
signature. All sanity checking is done when the payload is first
received, so the verification is complete once this point is reached.
The XML has also been added since a new config option was added to
general (curl_timeout). The maximum amount of time to wait for a
download can be configured through this option, with a low value by
default.
Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c
Some places in Asterisk did not treat the formats on a stream
as immutable when they are.
The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.
An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.
ASTERISK-28846
Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe
When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".
ASTERISK-28841
Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13
This unit test runs through combinations of...
* Local codecs
* Remote Codecs
* Codec Preference
* Incoming/Outgoing
A few new APIs were created to make it easier to test
the functionality but didn't result in any actual
functional change.
ASTERISK_28777
Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164
Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.
* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)
* Add "call_direction" to res_pjsip_session.
* Update pjsip_session_caps.c to make the functions more generic
so they could be used for both incoming and outgoing.
* Update ast_sip_session_create_outgoing to create the
pending_media_state->topology with the results of
ast_sip_session_create_joint_call_stream().
* The endpoint "preferred_codec_only" option now automatically sets
AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.
* A helper function ast_stream_get_format_count() was added to
streams to return the current count of formats.
ASTERISK-28777
Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
This change provides functions that take in a JSON payload, verify that
the contents contain all the mandatory fields and required values (if
any), and signs the payload with the private key. Four fields are added
to the payload: x5u, attest, iat, and origid. As of now, these are just
placeholder values that will be set to actual values once the logic is
implemented for what to do when an actual payload is received, but the
functions to add these values have all been implemented and are ready to
use. Upon successful signing and the addition of those four values, a
ast_stir_shaken_payload is returned, containing other useful information
such as the algorithm and signature.
Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393
named_acl.c (which is really a named_ha) now uses ast_ha_output.
I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set. If this works then we can add
similar to other modules as required.
Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f
This fixes ast_addressfamily_to_sockaddrsize to reference the
provided argument, and ast_sockaddr_from_sockaddr to not use the name of
a structure as argument.
Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7
This commit sets up some of the initial framework for the module and
adds a way to read the private key from the specified file, which will
then be appended to the certificate object. This works fine for now, but
eventually some other structure will likely need to be used to store all
this information. Similarly, the caller_id_number is specified on the
certificate config object, but in the end we will want that information
to be tied to the certificate itself and read it from there.
A method has been added that will retrieve the private key associated
with the caller_id_number passed in. Tab completion for certificates and
stores has also been added.
Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f
Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.
This patch does the following:
Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.
Adds a new ast_sip_session_caps structure that's set for each session media
object.
Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.
ASTERISK-28756 #close
Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.
Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.
ASTERISK-28755 #close
Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f
Update the state of remote_hold immediately on receipt of remote
SDP so that the information is available when building the SDP
answer
ASTERISK-28754 #close
Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f
There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.
Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e
There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.
Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113
This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.
This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.
ASTERISK-28697
Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
ASTERISK-28733
Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.
A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.
ASTERISK-28716
Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867
In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use. To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes" In Asterisk versions <18, the default
is "no" preserving existing behavior. Beginning with
Asterisk 18, the option will default to "yes".
NOTE: This change does not affect UserEvents or the ARI
TextMessageReceived events.
* Added the "hide_messaging_ami_events" option to asterisk.conf.
* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
the "Message/ast_msg_queue" channel if the option is set in
asterisk.conf. This suppresses the reporting of the events.
Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b
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
ast_addressfamily_to_sockaddrize will determine the size that's
required, and ast_sockaddr_from_sockaddr then wraps this new function
and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without
knowing the address family) into the ast_sockaddr structure.
Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
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
When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
When a topic is added to a pool, its name has the pool's topic
name prepended. For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.
When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.
Fix:
* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.
ASTERISK-28633
Reported by: Joeran Vinzens
Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6
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
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
ASTERISK-28604
Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
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
Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies. It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.
Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
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
* Pass caller information to frame allocation functions.
* Disable caching as it interfers with MALLOC_DEBUG reporting.
* Stop using ast_calloc_cache.
Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260
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
Serializer pools have previously existed in Asterisk. However, for the most
part the code has been duplicated across modules. This patch abstracts the
code into an 'ast_serializer_pool' object. As well the code is now centralized
in serializer.c/h.
In addition serializer pools can now optionally be monitored by a shutdown
group. This will prevent the pool from being destroyed until all serializers
have completed.
Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971
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
Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based
functions would result in the reference not being logged under
REF_DEBUG. This could sometimes cause inaccurate logging if NULL was
accidentally passed to a reference action. Now reference logging is
only disabled by option passed to the allocation method.
Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54
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
This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.
Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.
ASTERISK-28512
Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2
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
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 new function takes in a pointer to an ast_sockaddr structure,
a hostname and an optional port and then dispatches parallel
"AAAA" and "A" record queries. If an "AAAA" record is returned,
it's parsed into the ast_sockaddr structure along with the port
if it was supplied. If no "AAAA" record was returned, the
first "A" record returned (if any) is parsed instead.
This is a synchronous call. If you need asynchronous lookups,
use ast_dns_query_set_resolve_async and roll your own.
Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
When updating times on CDR or CEL records using the time at which
it is done can result in times being incorrect if the system is
heavily loaded and stasis message processing is delayed.
This change instead makes it so CDR and CEL use the time at which
the stasis messages that drive the systems are created. This allows
them to be backed up while still producing correct records.
ASTERISK-28498
Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a
When fixing ASTERISK~24212, a change was done so a scheduled callback could not
be removed while it was running. The caller of ast_sched_del would have to wait.
However, when the caller of ast_sched_del is the callback itself (however wrong
this might be), this new check would cause a deadlock: it would wait forever
for itself.
This changeset introduces an additional check: if ast_sched_del is called
by the callback itself, it is immediately rejected (along with an ERROR log and
a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the
after-ast_sched_del-refcall function is only run if ast_sched_del returned
success.
This should fix the following spurious race condition found in chan_sip:
- thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE)
- thread 2: run sip_poke_peer_now
- thread 2: blank out sched-ID (too soon!)
- thread 1: set sched-ID (too late!)
- thread 2: try to delete the currently running sched-ID
After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor
excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and
other madness) should occur.
(Thanks Richard Mudgett for reviewing/improving this "scary" change.)
Note that this change does not fix the observed race condition: unlocked
access to peer->pokeexpire (and potentially other scheduled items in chan_sip),
causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the
deadlock go away. And in the observed case, it will not have adverse affects
(like memory leaks) because the scheduled item is removed through a different
path.
ASTERISK-28282
Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856
** Note **
This patch is meant to be the minimum needed in order for the MWI core to use
the now underlying stasis_state module. As such it does not completely remove
its reliance on the stasis_cache. Doing so has allowed current consumers to
not have to change, and update those code paths for this patch. When time
allows, subsequent patches can/will be made to those consumers to take advantage
of some of the new MWI API included here. Thus, eventually and ultimately
removing MWI dependency on the stasis_cache.
** End Note **
This patch makes it so the MWI core now takes advantage of the new stasis_state
API. Consumers of MWI should no longer need to depend upon stasis topic pooling,
and the stasis cache directly. Similar functionality and implementation details
have now been pushed into the stasis_state module. However, all MWI state should
be accessed via the MWI API itself.
As such a few new methods, and constructs have been added to the MWI core that
facilitate consumer publishing, subscribing, and iterating over MWI state data.
* ast_mwi_subscriber *
Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox
in order to receive updates about the given mailbox. Adding a subscriber will
create the underlying topic, and associated state data if those do not already
exist for it. The topic, and last known state data is guaranteed to exist for
the lifetime of the subscriber.
* ast_mwi_publisher *
Before publishing to a particular topic a publisher should be created. This can
be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then
be done using one of the MWI publish functions. This ensures the message is
published to the appropriate topic, and the last known state is maintained.
* ast_mwi_observer *
Add an observer in order to watch for particular MWI module related events. For
instance if a submodule needs to know when a subscription is added to any
mailbox an observer can be added to watch for that.
* other *
Urgent message count is now part of the published MWI state object. Also state
can be iterated over using defined callbacks.
ASTERISK-28442
Change-Id: I93f935f9090cd5ddff6d4bc80ff90703c05cf776
This new module describes an API that can be thought of as a combination of
stasis topic pools, and caching. Except, hopefully done in a more efficient
and less memory "leaky" manner.
The API defines methods, and data structures for managing, and tracking
published message state through stasis. By adding a subscriber or publisher,
consumers can more easily track the lifetime of the contained state. For
instance, when no more publishers and/or subscribers have need of the topic,
and associated state its data is removed from the managed container.
* stasis_state_manager *
The manager stores and well, manages state data. Each state is an association
of a unique stasis topic, and the last known published stasis message on that
topic. There is only ever one managed state object per topic. For each topic
all messages are forwarded to an "all" topic also maintained by the manager.
* stasis_state_subscriber *
Topic and state can be created, or referenced within the manager by adding a
stasis_state_subscriber. When adding a subscriber if no state currently exists
new managed state is immediately created. If managed state already exists then
a new subscriber is created referencing that state. The managed state is
guaranteed to live throughout the subscriber's lifetime. State is only removed
from the manager when no other entities require it.
* stasis_state_publisher *
Topic and state can be created, or referenced within the manager by also adding
a stasis_state_publisher. When adding a publisher if no state currently exists
new managed state is created. If managed state already exists then a new
publisher is created referencing that state. The managed state is guaranteed to
live throughout the publisher's lifetime. State is only removed from the
manager when no other entities require it.
* stasis_state_observer *
Some modules may wish to watch for, and react to managed state events. By
registering a state observer, and implementing handlers for the desired
callbacks those modules can do so.
* other *
Callbacks also exist that allow consumers to iterate over all, or some of the
managed state.
ASTERISK-28442
Change-Id: I7a4a06685a96e511da9f5bd23f9601642d7bd8e5
When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.
Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1
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
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
Added a conversion for umax (largest maximum sized integer allowed). Adjusted
the other current conversion functions (uint and ulong) to be derivatives of
the umax conversion since they are simply subsets of umax.
Also made the negative check move the pointer on spaces since strtoumax does it
anyways.
Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08
After a bridge has been deleted the stasis control will depart
the channel and might attempt to re-add it to the dial bridge.
The later can fail and this can lead to a situation that the stasis
control is unlinked but the after_bridge_cb_failed cb is executed trying
to access a dangling control object.
Fix it by calling the after_cb's before bridge_channel_impart_signal.
ASTERISK-26718
Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496
When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.
This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.
ASTERISK-28401
Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c
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
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
ASTERISK-28363
Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1
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
Asterisk assumes that dlopen() will always run the constructor of a
shared library and every dlclose() will run its destructor. But dlopen()
may be permanent, meaning the constructor will only be run once, as is
the case with musl libc.
With a permanent dlopen() the Asterisk module loader does not work
correctly, because it's expectations regarding when the constructors and
destructors are run are not met. In fact a segmentation fault will occur
when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS
set (the dlopen() does not call the constructor, resource_being_loaded
is not set to NULL, then strlen is called with NULL instead of a string,
see issue ASTERISK-28319).
This commit adds code to the loader that will manually run the
constructors/destructors of the (non-builtin) modules where needed. To
achieve this a new ao2 container (linked list) is started and filled
with objects that contain the names of the modules and the pointers to
their respective info structs.
This behavior can be activated when configuring Asterisk
(--enable-permanent-dlopen). By default this is disabled, of course.
ASTERISK-28319 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875
Added topic_all container for centralizing the topic. This makes more
easier to managing the topics.
Added cli commands.
stasis show topics : It shows all registered topics.
stasis show topic <name> : It shows speicifed topic's detail info.
ASTERISK-28264
Change-Id: Ie86d125d2966f93de74ee00f47ae6fbc8c081c5f
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
Since the new names went in, the maximum taskprocessor name is too
short. This patch increases the name field to a length to better
handle the new names.
Change-Id: I32f32d6926f25c8ef5a91303fd2988d2c2858877
Added ability to specifiy a wizard is read-only when applying
it to a specific object type. This allows you to specify
create, update and delete callbacks for the wizard but limit
which object types can use them.
Added the ability to allow an object type to have multiple
wizards of the same type. This is indicated when a wizard
is added to a specific object type.
Added 3 new sorcery wizard functions:
* ast_sorcery_object_type_insert_wizard which does the same thing
as the existing ast_sorcery_insert_wizard_mapping function but
accepts the new read-only and allot-duplicates flags and also
returns the ast_sorcery_wizard structure used and it's internal
data structure. This allows immediate use of the wizard's
callbacks without having to register a "wizard mapped" observer.
* ast_sorcery_object_type_apply_wizard which does the same
thing as the existing ast_sorcery_apply_wizard_mapping function
but has the added capabilities of
ast_sorcery_object_type_insert_wizard.
* ast_sorcery_object_type_remove_wizard which removes a wizard
matching both its name and its original argument string.
* The original logic in __ast_sorcery_insert_wizard_mapping was moved
to __ast_sorcery_object_type_insert_wizard and enhanced for the
new capabilities, then __ast_sorcery_insert_wizard_mapping was
refactored to just call __ast_sorcery_insert_wizard_mapping.
* Added a unit test to test_sorcery.c to test the read-only
capability.
Change-Id: I40f35840252e4313d99e11dbd80e270a3aa10605
This might be useful in situations where you are loading an undetermined number
of items into a vector and don't want to keep (potentially) 2x the necessary
memory around indefinitely.
Change-Id: I9711daa0fe01783fc6f04c5710eba84f2676d7b9
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
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
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
Add a json_pack at startup that will fail if runtime links against a
library older than jansson-2.11.
Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a
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
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
The current settings AST_PBX_MAX_STACK is 128 entries which is
too low for some FreePBX installations with complex parking
arrangements. Increased to 512 if LOW_MEMORY is not defined.
ASTERISK-28300
Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299
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
Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a
little easier to retrieve a string field from the JSON object.
Also added an 'ast_strings_equal' function that safely checks (checks for NULLs)
for equality between two strings.
Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b
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
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
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
A bug in GCC causes TEST_CEL to return failure under the following
conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline. The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()
Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7
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
During Bridging of two channels if masquerade operation is performed on a
channel (clone channel) which was created with endpoint details
(ast_channel_alloc_with_endpoint()) and the original channel which is created
without endpoint details (ast_channel_alloc()) then both the channels must
exchange their endpoint details or else after masquerade when clone channel
is being destroyed the endpoint cleanup callbacks will be destroyed too and
after call completion unique_id of original channel will still be there in
ast_endpoint structure's channel_ids container.
ASTERISK-28197
Change-Id: I97ce73da390af20fd082fb09d722a6fe9cb2f39d
Ben Ford
Kevin Harwell
Sean Bright
George Joseph
Joshua C. Colp
Nathan Bruning
Jean Aunis
Jaco Kroon
Torrey Searle
Walter Doekes
Friendly Automation
Seán C McCord
Joshua Colp
Corey Farrell
Florian Floimair
Walter Doekes
Matt Jordan
Holger Hans Peter Freyther
Antoni Goldstein
Dan Cropp
Sebastian Kemper
sungtae kim
Matthew Fredrickson
Chris-Savinovich
mohitdhiman