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)
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.
Kevin Harwell [Tue, 17 Mar 2020 20:54:25 +0000 (15:54 -0500)]
ast_coredumper: add Asterisk information dump
This patch makes it so ast_coredumper now outputs the following information to
a *-info.txt file when processing a core file:
asterisk version and "built by" string
BUILD_OPTS
system start, and last reloaded date/time
taskprocessor list
equivalent of "bridge show all"
equivalent of "core show channels verbose"
Also a slight modification was made when trying to obtain the pid(s) of a
running Asterisk. If it fails to retrieve any it now reports an error.
Jaco Kroon [Fri, 20 Mar 2020 14:12:05 +0000 (16:12 +0200)]
netsock2: compile fixes.
This fixes ast_addressfamily_to_sockaddrsize to reference the
provided argument, and ast_sockaddr_from_sockaddr to not use the name of
a structure as argument.
Jaco Kroon [Wed, 18 Mar 2020 09:38:30 +0000 (11:38 +0200)]
dundi: fix NULL dereference.
If a negative (error) return is received from dundi_lookup_internal,
this is not handled correctly when assigning the result to the buffer.
As such, use a signed integer in the assignment and do a proper
comparison.
When the Asterisk receives 200 OK with invalid SDP,
the Asterisk/PJPROJECT terminating the session.
But if the channel was in the Bridge, Asterisk tries send
the Re-Invite before terminating the session.
And when the Asterisk sending the Re-Invite, it doesn't check
the SDP is NULL or not. This crashes the Asterisk.
Fixed it to close the session correctly if the UAS sends the
200 OK with wrong SDP.
Jaco Kroon [Wed, 18 Mar 2020 09:49:39 +0000 (11:49 +0200)]
build: enable building with uClibc
This patch has been included in Gentoo distribution for at least since
asterisk 1.8, but there are references in the logs going back as far as
1.0.0 - not sure if this is still required in any way, it does apply,
and it doesn't (as far as we can determine) cause build failures.
chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active
Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).
ASTERISK-28774 Reported-by: Michael Neuhauser
Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
Jaco Kroon [Wed, 27 Nov 2019 13:54:39 +0000 (15:54 +0200)]
res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses.
A pure blacklist is not good enough, we need a whitelist mechanism as
well, and the simplest way to do that is to re-use existing ACL
infrastructure.
This makes it simpler to blacklist say an entire block (/24) except a
smaller block (eg, a /29 or even a /32). Normally you'd need to
recursively split the block, so if you want to blacklist a /24 except
for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I
feel that having an ACL instead of a blacklist only is clearer.
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.
Joshua C. Colp [Thu, 12 Mar 2020 14:22:06 +0000 (11:22 -0300)]
audiohook: Don't allow audiohooks to attach to hung up channels.
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.
George Joseph [Wed, 4 Mar 2020 21:45:40 +0000 (14:45 -0700)]
CI: Create generic jenkinsfile
This is a generic jenkinsfile to build Asterisk and optionally
perform one or more of the following:
* Publish the API docs to the wiki
* Run the Unit tests
* Run Testsuite Tests
This job can be triggered manually from Jenkins or be triggered
automatically on a schedule based on a cron string.
Torrey Searle [Fri, 6 Mar 2020 16:13:34 +0000 (17:13 +0100)]
res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use
bridge_p2p_rtp_write will forward rtp to the bridged rtp instance
without modifying the ssrc. However, it is not updating the SSRC
in the bridged rtp. Thus, when SSRC packets are generated, they
have the correct SSRC for the sender.
Paulo Vicentini [Wed, 26 Feb 2020 00:30:04 +0000 (01:30 +0100)]
chan_pjsip: Check audio frame when remote SSRC changes.
If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.
The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.
Sean Bright [Fri, 6 Mar 2020 20:59:37 +0000 (15:59 -0500)]
enum.c: Add support for regular expression flag in NAPTR record
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.
res_rtp_asterisk: Add 'rtp show settings' cli command
This change introduce a CLI command for the RTP to display the general
configuration.
In the first step add the follow fields of the configurations:
- rtpstart
- rtpend
- dtmftimeout
- rtpchecksum
- strictrtp
- learning_min_sequential
- icesupport
Torrey Searle [Thu, 5 Mar 2020 09:08:54 +0000 (10:08 +0100)]
res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated
If ICE support is enabled but not negotiated, the rtp->ice structure is
not being destroyed. This leads to Asterisk waiting for ICE to complete
instead of immediately starting the DTLS handshake, resulting in the
call leg having no RTP.
Sean Bright [Wed, 4 Mar 2020 22:53:57 +0000 (17:53 -0500)]
enum.c: Make ast_get_txt() actually do something.
The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in 65b8381550a9f46fdce84de79960073e9d51b05d such that we would never
actually make a DNS TXT query as described.
Sebastian Kemper [Sun, 12 Jan 2020 11:37:46 +0000 (12:37 +0100)]
check_expr2: fix cross-compile/hardening issues
When building check_expr2 with ASLR PIE hardening enabled the linker
fails. This is resolved by adding the regular compiler flags when
building the object files from ast_expr2f.c and ast_expr2.c.
Note: The STANDALONE define is removed because it is already defined in
_ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives
'--enable-dev-mode'.
Also, a Makefile variable "CROSS_COMPILING" is added so that the
build system doesn't try to run check_expr2 when cross-compiling,
because that will fail the build as will.
ASTERISK-28685 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915
Kevin Harwell [Fri, 28 Feb 2020 18:53:40 +0000 (12:53 -0600)]
message & stasis/messaging: make text message variables work in ARI
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.
That being the case, and since this is technically an API breaking change (no
one should really be affected since things never really worked) the ARI version
was updated to reflect that.
Jaco Kroon [Thu, 20 Feb 2020 12:52:06 +0000 (14:52 +0200)]
addons/res_config_mysql: silense warnings about printf format errors.
Warnings without this:
res_config_mysql.c: In function 'update2_mysql':
res_config_mysql.c:741:15: warning: format '%llu' expects argument of type
'long long unsigned int', but argument 6 has type 'my_ulonglong'
{aka 'long unsigned int'} [-Wformat=]
ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n",
numrows, tablename);
Ben Ford [Thu, 23 Jan 2020 19:17:06 +0000 (13:17 -0600)]
RTP/ICE: Send on first valid pair.
When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.
A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.
Joshua C. Colp [Tue, 18 Feb 2020 13:10:16 +0000 (13:10 +0000)]
pjsip: Update ACLs on named ACL changes.
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.
Sean Bright [Mon, 17 Feb 2020 14:05:16 +0000 (09:05 -0500)]
app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
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.
Torrey Searle [Wed, 12 Feb 2020 16:05:11 +0000 (17:05 +0100)]
res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough
When moh_passthrough is used, asterisk is only generating invites
of type sendonly and sendrecv instead of taking fully into account
the on hold state of the local and remote parties
Sean Bright [Tue, 11 Feb 2020 13:46:02 +0000 (08:46 -0500)]
res_musiconhold: Avoid spurious warning when 'format' is the empty string
The change to res_config_odbc that allowed empty strings to be
returned to realtime consumers¹ causes a warning to be emitted when
loading MoH classes. So we need to treat an empty 'format' as if it
was not specified to avoid the warning.
Currently aptitude is installed using interactive mode. This patch
changes this to use the non-interactive mode as it can block
automatic dependencies installation, ex: CI, Docker build.
Walter Doekes [Tue, 28 Jan 2020 08:34:45 +0000 (09:34 +0100)]
chan_sip: Clarify in sample docs how directmediapermit/-acl should be used
It said "restrict [...] which peers should be able to pass [audio]
to each other".
However, these settings are not global (for which you would expect
signaling IPs to be checked). These settings are available per peer
only, and the IPs being checked, are the RTP IPs.
Sean Bright [Wed, 29 Jan 2020 14:57:38 +0000 (09:57 -0500)]
res_config_odbc: Preserve empty strings returned by the database
When res_config_odbc (and perhaps other realtime backends) reads a SQL
NULL from the database, it coalesces the value to the empty string
which prevents it from being returned to the realtime core.
However, if it instead reads the empty string from the database, it
needs a way to encode that fact without having the value omitted
entirely. It does this by changing the value to a string with a single
space. The realtime code in main/config.c recognizes this special case
and _turns the string back into the empty string_ before passing it to
realtime API consumers.
For all of this to work, we need to ensure that we actually pass the
single-space-string back to the realtime core, which is currently
failing because we are trimming the value before checking its
content. So instead we now special case the single-space-string case
so that empty values are returned properly.
Joshua C. Colp [Tue, 28 Jan 2020 15:18:45 +0000 (15:18 +0000)]
res_pjsip_pubsub: Increment persistence data ref when recreating.
Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.
Kevin Harwell [Mon, 27 Jan 2020 18:01:15 +0000 (12:01 -0600)]
stasis/app: don't lock an app before a call to send
Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.
This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.
Kevin Harwell [Mon, 27 Jan 2020 17:44:45 +0000 (11:44 -0600)]
res_stasis: trigger cleanup after update
The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.
This patch makes it so the cleanup code is executed post 'update'.
George Joseph [Wed, 22 Jan 2020 18:56:38 +0000 (11:56 -0700)]
cdr.c: Set event time on party b when leaving a parking bridge
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.
Sean Bright [Wed, 22 Jan 2020 15:39:47 +0000 (10:39 -0500)]
http: Add ability to disable /httpstatus URI
Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.
We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.
Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.
Additionally:
* Change 'enablestatic' to 'enable_static' but keep the former for
backwards compatibility.
* Improve some internal variable names
projects / thirdparty / asterisk.git / log
