Walter Doekes [Tue, 16 Jun 2020 13:18:11 +0000 (15:18 +0200)]
app_queue: Read latest wrapuptime instead of (possibly stale) copy
Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.
This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.
In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.
Kevin Harwell [Fri, 15 May 2020 21:08:20 +0000 (16:08 -0500)]
pjproject: Upgrade bundled version to pjproject 2.10
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.
Joshua C. Colp [Wed, 3 Jun 2020 16:47:42 +0000 (13:47 -0300)]
core_unreal / core_local: Add multistream and re-negotiation.
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.
Joshua C. Colp [Mon, 8 Jun 2020 11:27:53 +0000 (08:27 -0300)]
res_rtp_asterisk: Don't assume setting retrans props means to enable.
The "value" passed in when setting an RTP property determines
whether it should be enabled or disabled. The RTP send and
receive retrans props did not examine this to know if the
buffers should be enabled. They assumed they always should be.
This change makes it so that the "value" passed in is
respected.
Joshua C. Colp [Wed, 10 Jun 2020 17:11:16 +0000 (14:11 -0300)]
bridge_softmix: Add additional old states for adding new source.
There are three states that an old stream can be in to allow
becoming a source stream in a new stream:
1. Removed
2. Inactive
3. Sendonly
This change adds the two missing ones, inactive and sendonly,
so if a stream transitions from those to a state where they are
providing video to Asterisk we properly re-negotiate the other
participants.
George Joseph [Wed, 3 Jun 2020 16:23:31 +0000 (10:23 -0600)]
res_fax: Don't start a gateway if either channel is hung up
When fax_gateway_framehook is called and a gateway hasn't already
been started, the framehook gets the t38 state for both the current
channel and the peer. That call trickles down to the channel
driver which determines the state. If either channel is hung up
(or in the process of being hung up), the channel driver's tech_pvt
is going to be NULL which, in the case of chan_pjsip, will cause a
segfault.
* Added a hangup check for both the channel and peer channel
before starting a fax gateway.
* Added a check for NULL tech_pvt to chan_pjsip_queryoption
so we don't attempt to reference a tech_pvt that's already
gone.
George Joseph [Mon, 8 Jun 2020 00:02:00 +0000 (18:02 -0600)]
app_confbridge: Plug ref leak of bridge channel with send_events
When send_events is enabled for a user, we were leaking a reference
to the bridge channel in confbridge_manager.c:send_message(). This
also caused the bridge snapshot to not be destroyed.
Kevin Harwell [Mon, 1 Jun 2020 23:25:48 +0000 (18:25 -0500)]
Compiler fixes for gcc 10
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.
Joshua C. Colp [Fri, 5 Jun 2020 09:45:18 +0000 (06:45 -0300)]
bridge_channel: Don't queue unmapped frames.
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.
Joshua C. Colp [Wed, 27 May 2020 08:47:07 +0000 (05:47 -0300)]
res_fax: Don't consume frames given to fax gateway on write.
In a particular fax gateway scenario whereby it would
have to translate using the read translation path on a
channel the frame being translated would be consumed.
When the frame is in the write path it is not permitted
to free the frame as the caller expects it to continue
to exist.
This change makes it so that the frame is only consumed
on the read path where it is acceptable to free it.
Pirmin Walthert [Thu, 4 Jun 2020 06:50:35 +0000 (08:50 +0200)]
res_pjsip_logger: use the correct pointer when logging tx_messages to pcap
When writing tx messages to pcap files, Asterisk is using the wrong
pointer resulting in lots of wasted space. This patch fixes it to use
the correct pointer.
George Joseph [Thu, 14 May 2020 18:24:19 +0000 (12:24 -0600)]
Scope Tracing: A new facility for tracing scope enter/exit
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.
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.
Pirmin Walthert [Fri, 29 May 2020 09:28:57 +0000 (11:28 +0200)]
res_pjsip_logger.c: correct the return value checks when writing to pcap
files
fwrite() does return the number of elements written and not the
number of bytes. However asterisk is currently comparing the return
value to the size of the written element what means that asterisk logs
five WARNING messages on every packet written to the pcap file.
This patch changes the code to check for the correct value, which will
always be 1.
Joshua C. Colp [Wed, 27 May 2020 14:35:42 +0000 (11:35 -0300)]
res_pjsip: Use correct pool for storing the contact_user value.
When replacing the user portion of the Contact URI the code
was using the ephemeral pool instead of the tdata pool. This
could cause the Contact user value to become invalid after a
period of time.
The code will now use the tdata pool which persists for the
lifetime of the message instead.
Pirmin Walthert [Wed, 13 May 2020 12:06:19 +0000 (14:06 +0200)]
res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header
While asterisk is filtering out the x-ast-orig-host parameter from the
contact on response messages, it is not filtering it out from the
request URI and the to header on SIP requests (for example INVITE).
Joshua C. Colp [Mon, 18 May 2020 14:05:56 +0000 (11:05 -0300)]
bridge: Don't try to match audio formats.
When bridging channels we were trying to match the audio
formats of both sides in combination with the configured
formats. While this is allowed in SDP in practice this
causes extra reinvites and problems. This change ensures
that audio streams use the formats of the first existing
active audio stream. It is only when other stream types
(like video) exist that this will result in re-negotiation
occurring for those streams only.
Alexander Traud [Mon, 18 May 2020 15:10:01 +0000 (17:10 +0200)]
res_srtp: Set all possible flags while selecting the Crypto Suite.
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.
Joshua C. Colp [Tue, 19 May 2020 12:55:32 +0000 (09:55 -0300)]
res_sorcery_config: Always reload configuration on errors.
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.
Joshua C. Colp [Tue, 19 May 2020 09:18:58 +0000 (06:18 -0300)]
bridge_softmix: Always remove audio from mixed frame.
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.
res_pjsip_logger: Expand functionality to improve logging.
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.
res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings
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
Joshua C. Colp [Tue, 12 May 2020 23:15:41 +0000 (20:15 -0300)]
ari: Allow variables to be set on channel create.
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.
Roger James [Sun, 10 May 2020 10:01:56 +0000 (11:01 +0100)]
pjsip_resolver.c: Ensure AAAA dns requests are made.
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)
Guido Falsi [Fri, 8 May 2020 11:11:47 +0000 (13:11 +0200)]
pjproject: Fix race condition when building with parallel make
Pjproject makefiles miss some dependencies which can cause race
conditions when building with parallel make processes. This patch
adds such dependencies correctly.
app.c: make sure that no non-async-signal-safe syscalls are used after
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.
George Joseph [Mon, 4 May 2020 16:31:57 +0000 (10:31 -0600)]
streams: Fix one memory leak and one formats ref issue
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.
Guido Falsi [Sun, 3 May 2020 10:30:15 +0000 (12:30 +0200)]
pjproject: Remove bashism from configure.m4 script
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
George Joseph [Thu, 30 Apr 2020 15:56:03 +0000 (09:56 -0600)]
app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict
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
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.
Alexander Traud [Mon, 27 Apr 2020 15:28:01 +0000 (17:28 +0200)]
core_local: Local calls are always secure.
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.
res_rtp_asterisk: Protect access to nochecksums with #ifdef
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.
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.
Peter Turczak [Fri, 17 Apr 2020 07:39:09 +0000 (08:39 +0100)]
chan_mobile: Add smoother to make SIP/RTP endpoints happy.
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.
Alexander Traud [Wed, 22 Apr 2020 17:38:13 +0000 (19:38 +0200)]
app_fax: SpanDSP headers do not use ast_malloc; ignore that.
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.
stream: Enforce formats immutability and ensure formats exist.
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.
fax: Fix crashes in PJSIP re-negotiation scenarios.
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.
confbridge: Add support for disabling text messaging.
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".
res_rtp_asterisk: Resolve loop when receive buffer is flushed
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.
res_rtp_asterisk: Free payload when error on insertion to data buffer
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.
Jaco Kroon [Tue, 3 Dec 2019 18:35:20 +0000 (20:35 +0200)]
res_rtp_asterisk: iterate all local addresses looking to populate ICE.
By using pjproject to give us a list of candidates, and then filtering,
if the host has >32 addresses configured, then it is possible that we
end up filtering out all 32 of those, and ending up with no candidates
at all. Instead, get getifaddrs (which pjsip is using underlying
anyway) to retrieve all local addresses, and iterate those, adding the
first 32 addresses not excluded by the ICE ACL.
In our setup at any point in time We've got between 6 and 328 addresses
on any given system. The lower limit is the lower limit but the upper
limit is growing on a near daily basis currently.
Alexander Traud [Sun, 12 Apr 2020 14:53:50 +0000 (16:53 +0200)]
BuildSystem: Search for Python/C API when possibly needed only.
The Python/C API is used only if the Test Framework was enabled in Asterisk
'make menuselect'. The Test Framework is available only if the Developer Mode
was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API
is used only if the PJProject was found and not disabled in Asterisk; the user
did not go for './configure --without-pjproject'.
Furthermore, because version 2 of that Python/C API is required (currently) and
because some platforms do not offer a generic version 2, the script searches
for 2.7 explicitly as well.
To avoid version mismatch between the Python/C API and the Python environment,
the script searches for the latter in the same versions, in the same the order
as well. Because this Python/C API is just for (some) Asterisk contributors,
the script also goes for the Python 3 environment as a last resort for all
other Asterisk users. This allows 'make full' even on minimal installations of
Ubuntu 18.04 LTS and newer.
Because the Python/C API is Asterisk contributor specific, the Python packages
are removed from the script './contrib/scripts/install_prereq' as this script
is intended for Asterisk users. Asterisk contributors have to install much more
packages in any case, like:
sudo apt install autoconf automake git git-review python2.7-dev
It is possible to configure a TCP/TLS client without having a TCP/TLS
server. In that case, no error or warning was printed but the headers
Contact and Via in SIP REGISTER were "(null)".
Kevin Harwell [Wed, 8 Apr 2020 19:33:47 +0000 (14:33 -0500)]
chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet
If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.
This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.
This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.
Joshua C. Colp [Thu, 26 Mar 2020 22:42:27 +0000 (19:42 -0300)]
res_pjsip: Don't set endpoint to unavailable in all cases.
When an AOR is modified endpoints are updated that reference
the AOR so they can start receiving updates and reflect the
correct state. If this is the case then we shouldn't change
the endpoint to be offline if it does not reference the AOR
but instead only when the endpoint is completely updated for
all its AORs.
ASTERISK-28056
patches:
pjsip_options-aor.diff submitted by jhord (license 6978)
Kevin Harwell [Tue, 31 Mar 2020 17:52:44 +0000 (12:52 -0500)]
channel: write to a stream on multi-frame writes
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.
sungtae kim [Thu, 26 Mar 2020 22:18:17 +0000 (22:18 +0000)]
dial.c: Removed dial string 80 character limitation
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.
Jaco Kroon [Wed, 18 Mar 2020 13:49:56 +0000 (15:49 +0200)]
acl: implement a centralized ACL output mechanism for HAs and ACLs.
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.
When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.
This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.
projects / thirdparty / asterisk.git / log
