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
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
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
If a frame is written to a channel in a bridge we
would normally queue this frame up and the channel
thread would then act upon it. If this frame had no
stream mapping on the channel it would then be
discarded.
This change adds a check before the queueing occurs
to determine if a mapping exists. If it does not
exist then the frame is not even queued at all. This
stops a frame duplication from happening and from
the channel thread having to wake up and deal with
it.
Change-Id: I17189b9b1dec45fc7e4490e8081d444a25a00bda
If the bridge show all command could not get the bridge snapshot, it causes null pointer exception.
Fixed it to check the snapshot is null.
ASTERISK-28920
Change-Id: I3521fc1b832bfc69644d0833f2c78177e1e51f58
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
fork before exec
Posix does only allow async-signal-safe syscalls after fork before exec.
As asterisk ignores this, functions like TrySystem or System sometimes
end up in a deadlocked child process. The patch prevents the use of
non-async-signal-safe syscalls.
ASTERISK-28776
Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e
ast_stream_topology_create_from_format_cap() was setting the
stream->formats directly but not freeing the default formats. This
causes a memory leak.
* ast_stream_topology_create_from_format_cap() now calls
ast_stream_set_formats() which properly cleans up the existing
stream formats.
When cloning a stream, the source stream's format caps _pointer_ is
copied to the new stream and it's reference count bumped. If
either stream is set to "removed", this will cause _both_ streams
to have their format caps cleared.
* ast_stream_clone() now creates a new format caps object and copies
the formats from the source stream instead of just copying the
pointer.
ASTERISK-28870
Change-Id: If697d81c3658eb7baeea6dab413b13423938fb53
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
In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support
the channel items 'secure_bridge_media' and 'secure_bridge_signaling'.
That way, a channel can be forced to use encryption even if not
specified in its configuration.
However, when the Local Proxy kicks in, for example, in case of a
forwarding (SIP status 302), Local Proxy stated it does not know those
items. Consequently, such a call could not be proxied how clever your
Dialplan was. Because local calls within Asterisk are always secure,
Local Proxy accepts such a request now.
ASTERISK-22920
Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c
While testing the latest RC on FreeBSD I noticed this new file fails to build. On FreeBSD inlcuding resolv.h requires sockaddr_in to be defined, and it's defined in netinet/in.h. So I added this include.
ASTERISK-28853 #close
Change-Id: I6997daf3956e6eb70ab6cb358628d162fad80079
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
When the ast_data_buffer_put rejects to add a packet, for example because
the buffer already contains a packet with the same sequence number, the
payload will never be freed, resulting in a memory leak.
The data buffer will now return an error if this situation occurs
allowing the caller to free the payload. The res_rtp_asterisk module
has also been updated to do this.
ASTERISK-28826
Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1
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
If a frame handling routine returns a list of frames (vs. a single frame)
those frames are never passed to a tech's write_stream handler even if one is
available. For instance, if a codec translation occurred and that codec
returned multiple frames then those particular frames were always only sent
to the tech's "write" handler. If that tech (pjsip for example) was stream
capable then those frames were essentially ignored. Thus resulting in bad
audio.
This patch makes it so the "write_stream" handler is appropriately called
for all cases, and for all frames if available.
ASTERISK-28795 #close
Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d
The dial application had 80 characters of destination length
limitation. But this limitation causes unexpected dial string
cut if the dial string is long.
Removed unnecessary limited buffer to support longer dial
destination.
ASTERISK-27946
Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330
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
binutils 2.34 merged this commit:
https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\
h=fd3619828e94a24a92cddec42cbc0ab33352eeb4
Which effectively does things like:
-#define bfd_section_size(bfd, ptr) ((ptr)->size)
-#define bfd_get_section_size(ptr) ((ptr)->size)
+#define bfd_section_size(sec) ((sec)->size)
So in order to remain backwards compatible we need to detect this API
change, and adjust accordingly. The simplest is to notice that the
bfd_get_section_size and bfd_get_section_vma MACROs are no longer
defined, and define then onto the new API. The alternative is to litter
the code with a number of #ifdef #else #endif splatters right through
the code.
Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f
Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.
This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.
ASTERISK-28780
Change-Id: I8044c06daa06f0f16607788c596f55623be26f58
A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.
Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.
ASTERISK-26711 #close
Reported by: Vitold
Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0
The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in
65b8381550 such that we would never
actually make a DNS TXT query as described.
This patch restores the documented behavior.
ASTERISK-19460 #close
Reported by: George Joseph
Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376
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
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
Dump OpenSSL's error stack to the error log when things fail.
ASTERISK-28750 #close
Reported by: Martin Zeh
Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a
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 opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.
* If the user wants to write to a wav49 file, make sure that it is
reflected properly in MIXMONITOR_FILENAME.
* Add a note to the documentation describing this behavior.
* Add a note in main/file.c indicating that app_mixmonitor needs to be
changed if the logic in build_filename was changed.
ASTERISK-24798 #close
Reported by: xrobau
Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c
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
When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.
* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.
NOTE: This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.
ASTERISK-28677
Reported by: Maciej Michno
Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca
