Joshua C. Colp [Fri, 28 Aug 2020 09:29:23 +0000 (06:29 -0300)]
parking: Copy parker UUID as well.
When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.
Joshua C. Colp [Thu, 27 Aug 2020 10:31:40 +0000 (07:31 -0300)]
pbx: Fix hints deadlock between reload and ExtensionState.
When the ExtensionState AMI action is executed on a pattern matched
hint it can end up adding a new hint if one does not already exist.
This results in a locking order of contexts -> hints -> contexts.
If at the same time a reload is occurring and adding its own hint
it will have a locking order of hints -> contexts.
This results in a deadlock as one thread wants a lock on contexts
that the other has, and the other thread wants a lock on hints
that the other has.
This change enforces a hints -> contexts locking order by explicitly
locking hints in the places where a hint is added when queried for.
This matches the order seen through normal adding of hints.
Sean Bright [Thu, 6 Aug 2020 16:41:33 +0000 (12:41 -0400)]
res_musiconhold.c: Prevent crash with realtime MoH
The MoH class internal file vector is potentially being manipulated by
multiple threads at the same time without sufficient locking. Switch to
a reference counted list and operate on copies where necessary.
Sean Bright [Tue, 4 Aug 2020 15:51:16 +0000 (11:51 -0400)]
vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
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.
pjproject: clone sdp to protect against (nat) modifications
PJSIP, UDP transport with external_media_address and session timers
enabled. Connected to SIP server that is not in local net. Asterisk
initiated the connection and is refreshing the session after 150s
(timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has
a malformed IP address in its SDP (garbage string). This only happens
when the SDP is modified by the nat-code to replace the local IP address
with the configured external_media_address.
Analysis: the code to modify the SDP (in
res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?)
in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses
the tdata->pool to allocate the replacement string. But the same
pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also
used for the 2nd refresh-INVITE (because it is stored in pjmedia's
pjmedia_sdp_neg structure). The problem is, that at that moment, the
tdata->pool that holds the stringified external_media_address from the
1. refresh-INVITE has long been reused for something else.
Fix by Sauw Ming of pjproject (see
https://github.com/pjsip/pjproject/pull/2476): the local, potentially
modified pjmedia_sdp_stream is cloned in
pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the
clone is stored, thereby detaching from the tdata->pool (which is only
released *after* process_answer())
ASTERISK-28973 Reported-by: Michael Neuhauser
Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e
George Joseph [Thu, 23 Jul 2020 19:47:25 +0000 (13:47 -0600)]
res_pjsip_session: Ensure reused streams have correct bundle group
When a bundled stream is removed, its bundle_group is reset to -1.
If that stream is later reused, the bundle parameters on session
media need to be reset correctly it could mistakenly be rebundled
with a stream that was removed and never reused. Since the removed
stream has no rtp instance, a crash will result.
Sean Bright [Mon, 13 Jul 2020 20:06:14 +0000 (16:06 -0400)]
utf8.c: Add UTF-8 validation and utility functions
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.
res_pjsip_registrar: Don't specify an expiration for static contacts.
Statically configured contacts on an AOR don't have an expiration
time so when adding them to the resulting 200 OK if an endpoint
registers ensure they are marked as such.
When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.
The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.
pjsip: Include timer patch to prevent cancelling timer 0.
I noticed this while looking at another issue and brought
it up with Teluu. It was possible for an uninitialized timer
to be cancelled, resulting in the invalid timer id of 0
being placed into the timer heap causing issues.
This change is a backport from the pjproject repository
preventing this from happening.
res_http_websocket: Avoid reading past end of string
We read beyond the end of the buffer when copying the string out of the
buffer when we used ast_copy_string() because the original string was
not null terminated. Instead switch to ast_strndup() which does not
exhibit the same behavior.
George Joseph [Mon, 6 Jul 2020 15:57:18 +0000 (09:57 -0600)]
frame.c: Make debugging easier
* 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.
George Joseph [Sun, 5 Jul 2020 23:51:04 +0000 (17:51 -0600)]
Scope Trace: Make it easier to trace through synchronous tasks
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.
Kevin Harwell [Tue, 30 Jun 2020 16:08:47 +0000 (11:08 -0500)]
PJSIP_MEDIA_OFFER: override configuration on refresh
When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.
This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.
George Joseph [Tue, 30 Jun 2020 13:56:34 +0000 (07:56 -0600)]
Scope Trace: Add some new tracing macros and an ast_str helper
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.
Joshua C. Colp [Fri, 26 Jun 2020 10:18:55 +0000 (07:18 -0300)]
res_pjsip: Apply AOR outbound proxy to static contacts.
The outbound proxy for an AOR was not being applied to
any statically configured Contacts. This resulted in the
OPTIONS requests being sent to the wrong target.
This change sets the outbound proxy on statically configured
contacts once the AOR configuration is done being
applied.
Joshua C. Colp [Wed, 24 Jun 2020 10:25:47 +0000 (07:25 -0300)]
menuselect: Resolve infinite loop in dependency scenario.
Given a scenario where a module has a dependency on both
an external library and a module if the external library was
available and the module was not an infinite loop would
occur. This happened due to the code changing the dependecy
status to no failure on each dependency checking loop
iteration, resulting in the code thinking that it had
gone from no failure to failure each time triggering another
dependency check.
This change makes it so that the old dependency status is
preserved throughout the dependency checking allowing it to
determine that after the first iteration the dependency
status does not transition from no failure to failure.
Kevin Harwell [Mon, 22 Jun 2020 20:27:32 +0000 (15:27 -0500)]
chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+
A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.
This patch removes the reference, and instead initializes the value
to '0'.
Frederic LE FOLL [Mon, 22 Jun 2020 09:08:47 +0000 (11:08 +0200)]
chan_sip: chan_sip does not process 400 response to an INVITE.
chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".
According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
and handle_response().
- 414/493 only in handle_response_invite().
This fix adds 400 response support in handle_response_invite().
res_corosync: Fix crash in huge distributed environment.
1) Fix memory-leaks
Added code to release ast_events extracted from corosync and stasis messages
2) Clean stasis cache when a member of the corosync cluster leaves the group
Added code to remove from the stasis cache of the members remained on the
group all the messages with the EID of the left member.
If the device states of the left member remain in the stasis cache of other
members, they will not be updated anymore and high priority cached values,
like BUSY, will take precedence over current device states.
3) Stop corosync event propagation when node is not joined to the group
Updated dispatch_thread_handler code to detect when asterisk is not joined
to the corosync group and added some condition in publish_event_to_corosync
code to send corosync messages only when joined.
When a node is not joined its corosync daemon can't send messages:
the cpg_mcast_joined function append new messages to the FIFO buffer until
it's full and then it blocks indefinitely.
In this scenario if the stasis_message_cb callback, registered by
res_corosync to handle stasis messages, try to send a corosync messages,
the thread of the stasis thread-pool will be blocked until the node join
the corosync cluster.
ASTERISK-28888
Reported by: Università di Bologna - CESIA VoIP
Guido Falsi [Thu, 18 Jun 2020 10:14:26 +0000 (12:14 +0200)]
chan_dadhi: Fix setvar in dahdi channels
The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.
Joshua C. Colp [Thu, 18 Jun 2020 08:49:37 +0000 (05:49 -0300)]
app_stream_echo: Fix state of added streams.
When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.
This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.
Joshua C. Colp [Wed, 17 Jun 2020 08:58:44 +0000 (05:58 -0300)]
res_pjsip_session: Preserve label on incoming re-INVITE.
When a re-INVITE is received we create a new set of
streams that are then swapped in as the active streams.
We did not preserve the SDP label from the previous
streams, resulting in the label getting lost.
This change ensures that if an SDP label is present
on the previous stream then it is set on the new stream.
Joshua C. Colp [Wed, 10 Jun 2020 09:35:50 +0000 (06:35 -0300)]
res_sorcery_memory_cache: Disallow per-object expire with full backend.
The AMI action and CLI command did not take into account the properties
of full backend caching. This resulted in an expired object remaining
removed until a full backend update occurred, instead of having the
object updated when needed.
This change makes it so that the AMI action and CLI command for object
expire will now fail instead of putting the cache into an undesired
state. If full backend caching is enabled then only operations
which act on the entire cache are available.
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".
projects / thirdparty / asterisk.git / log
