When a configuration file in Asterisk is loaded
information about it is stored such that on a
reload it is not reloaded if nothing has changed.
This can be problematic when an error exists in
a configuration file in PJSIP since the error
will be output at start and not subsequently on
reload if the file is unchanged.
This change makes it so that if an error is
encountered when res_sorcery_config is loading
a configuration file a reload will always read
in the configuration file, allowing the error
to be seen easier.
Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed
The flags of a previous selection could have been set within the
object 'srtp', for example, when the previous selection returned
failure after setting just 'some' flags. Now, not to clutter the
code, all possible flags are cleared first, and then the selected
flags are set as before.
ASTERISK-28903
Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee
When receiving audio from a channel we determine if it
is talking or silence based on a threshold value. If
this threshold is met we always mix the audio into the
conference bridge. If this threshold is not met we also
mix the audio into the conference bridge UNLESS the
drop silence option is enabled.
The code that removed the audio from the mixed frame
assumed that it was always not present if it did not
meet the threshold to be considered talking. This is
incorrect. If it has been stated that the audio was
mixed into the mixed frame then it has been mixed into
the mixed frame. By not removing audio that was
considered non-talking it was possible for a channel
to receive a slight echo of audio of itself at times.
This change ensures that the audio is always removed
from the mixed frame going back to the channel so it
no longer receives the slight echo.
ASTERISK-28898
Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb
Added two unit tests, one for signing and another for verifying.
stir_shaken_sign checks to make sure that all the required parameters
are passed in and then signs the actual payload. If a signature is
produced and a payload returned as a result, the test passes.
stir_shaken_verify takes the signature from a signed payload to verify.
This unit test also verifies that all the required information is passed
in, and then attempts to verify the signature. If verification is
successful and a payload is returned, the test passes.
Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13
The PJSIP packet logger now has the following CLI commands:
pjsip set logger pcap <filename>
When used this will create a pcap file containing the incoming
and outgoing SIP packets, in unencrypted form.
pjsip set logger verbose <on / off>
This allows you to toggle logging to verbose on and off.
pjsip set logger host <IP/subnet mask> add
This allows you to add an additional IP address or subnet
mask to logging, allowing you to log multiple instead of
just a single IP address or all traffic.
The normal "pjsip set logger host" CLI command has also been
expanded to allow subnet masks as well.
ASTERISK-28895
Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e
A warning was triggered that there may be a problem regarding file
extension (which is correct and should not be set anyway). The warning
also appeared if there was dot within the path itself.
E.g.
[sales-queue-hold]
mode=playlist
entry=/var/www/domain.tld/moh/funky_music
The music played correctly but you get a warning message.
Now there will be a check if the position of a potential dot character
is after the last position of a slash character. This dot charachter
will be treated as a extension naming. Dots within the path then ignored.
ASTERISK-28892
Reported-By: Nicholas John Koch
Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e
This change adds the same variable functionality that
is available for originating a channel to the create
call. Now when creating a channel you can specify
dialplan variables to set instead of having to do another
API call.
ASTERISK-28896
Change-Id: If13997ba818136d7c070585504fc4164378aa992
1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups
when an IPV6 transport type has been requested.
2. Rename all occurrences of pjsip_transport_get_type_name to
pjsip_transport_get_type_desc. This ensures that the log/debug info
shows whether the transport is IPv6 or IPv4.
3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport
types. This results in invalid values. Use a bitwise or instead.
ASTERISK-26780
Patches:
pjsip_resolver.c uploaded by Peter Sokolov (License #7070)
Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58
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
Pjproject makefiles miss some dependencies which can cause race
conditions when building with parallel make processes. This patch
adds such dependencies correctly.
ASTERISK-28879 #close
Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru>
Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22
Changed source and destination address fields in struct
pjsip_history_entry so that they are long enough to hold an IPv6
address.
ASTERISK-28854
Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e
Ensure that output buffers for the osp_convert_inout
function have sufficient space for additional data
such as brackets and ports.
ASTERISK-28804
Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b
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
The configure.m4 script for pjproject contains some += syntax, which
is specific to bash, replacing it with string substitutions makes
the script compatible with traditional Bourne shells.
ASTERISK-28866 #close
Reported-by: Christoph Moench-Tegeder <cmt@FreeBSD.org>
Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05
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
The gcc 10 -Wrestrict option was causing "overlap" errors when
snprintf was copying one char[256] structure member to another
char[256] member in the same structure.
Using ast_alloca instead of declaring the structure inline
solves the issue.
Here's a link to the "enhancement":
https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html
We may follow up with a gcc bug report.
Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534
In practice it has been seen that some users come
close to our maximum ICE candidate count of 32.
In case people have gone over this increases the
count to 64, giving ample room.
ASTERISK-28859
Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf
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
Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined.
ASTERISK-28852 #close
Change-Id: I381718893b80599ab8635f2b594a10c1000d595e
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
In contrast to RFC 3551, section 4.2, several SIP/RTP clients misbehave
severly (up to crashing). This patch adds another smoother for the audio
received via bt. Therefore the audio frames sent to the core will be
CHANNEL_FRAME_SIZE.
ASTERISK-28832 #close
Change-Id: Ic5f9e2f35868ae59cc9356afbd1388b779a1267f
Since Asterisk 14, app_fax did not compile at all because Asterisk
requires that not malloc but ast_malloc is used everywhere. However,
the system headers of SpanDSP use malloc. Because we cannot (and do
not need to) change system headers, let us ignore this.
ASTERISK-28848
Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546
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
Fixed it to copy the entire string from the requested endpoint body except tech-prefix.
ASTERISK-28847
Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25
This change fixes a few re-negotiation issues
uncovered with fax.
1. The fax support uses its own mechanism for
re-negotiation by conveying T.38 information in
its own frames. The new support for re-negotiating
when adding/removing/changing streams was also
being triggered for this causing multiple re-INVITEs.
The new support will no longer trigger when
transitioning between fax.
2. In off-nominal re-negotiation cases it was
possible for some state information to be left
over and used by the next re-negotiation. This
is now cleared.
ASTERISK-28811
ASTERISK-28839
Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2
MODULEINFO is checked while buidling/linking the module.
AST_MODULE_INFO is checked while loading/running the module.
ASTERISK-28838
Change-Id: I55dc05ce19552d0415c9045021b42bd82ef44e52
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
MODULEINFO is checked while buidling/linking the module.
AST_MODULE_INFO is checked while loading/running the module.
ASTERISK-28838
Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042
When the receive buffer was flushed by a received packet while it
already contained a packet with the same sequence number, Asterisk
never left the while loop which tried to order the packets.
This change makes it so if the packet is in the receive buffer it
is retrieved and freed allowing the buffer to empty.
ASTERISK-28827
Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3
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
