minargs enables enforcing of minimum count of arguments to pass to
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
this should be set to 4. func_odbc will generate an error in this case,
so for example
[FOO]
minargs = 4
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
potentially leaked ARG4 from Gosub().
ARGC is needed if you're using optional argument, to verify whether or
not an argument has been passed, else it's possible to use a leaked ARGn
from Gosub (app_stack). So now you can safely do
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext. You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.
* Added logic to iax2_call() and authenticate_reply() to print
a warning and hanhup the call if encryption is requested and
there's no secret or auth method. This prevents the crash.
* Added the ability to specify a default "auth" in the [general]
section of iax.conf.
ASTERISK-29624
Reported by: N A
Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254
As described in the issue, /tmp is not a suitable location for a
large amount of cached media files, since most distributions make
/tmp a RAM-based tmpfs mount with limited capacity.
I opted for a location that can be configured separately, as opposed
to using a subdirectory of spooldir, given the different storage
profile (transient files vs files that might stay there indefinitely).
This commit just makes the cache directory configurable, and changes
the default location from /tmp to /var/cache/asterisk.
ASTERISK-29143
Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d
Currently any response from res_curl where we get an answer from the
web server, regardless of what the response is (404, 403 etc.) Asterisk
currently treats it as a success. This patch allows you to set which
codes should be considered as a failure by Asterisk. If say we set
failurecodes=404,403 then when using curl in realtime if a server gives
a 404 error Asterisk will try to failover to the next option set in
extconfig.conf
ASTERISK-28825
Reported by: Dovid Bender
Code by: Gobinda Paul
Change-Id: I94443e508343e0a3e535e51ea6e0562767639987
This changes the outgoing offer call preference
default option to match the behavior of previous
versions of Asterisk.
The additional advanced codec negotiation options
have also been removed from the sample configuration
and marked as reserved for future functionality in
XML documentation.
The codec preference options have also been fixed to
enforce local codec configuration.
ASTERISK-29109
Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2
app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge. To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second. The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".
Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a
Since ASTERISK_27978 the default is not off but 90 seconds. That change
happened because ASTERISK_27347 disabled the keep-alives in the bundled
PJProject and Asterisk should behave the same as before.
Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46
Added a new log formatter called "plain" that always prints
file, function and line number if available (even for verbose
messages) and never prints color control characters. It also
doesn't apply any special formatting for verbose messages.
Most suitable for file output but can be used for other channels
as well.
You use it in logger.conf like so:
debug => [plain]debug
console => [plain]error,warning,debug,notice,pjsip_history
messages => [plain]warning,error,verbose
Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d
This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.
Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d
Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.
Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.
Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.
Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1
If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.
(Actually, commit ed615afb7e already
partially fixed it for shared_lastcall.)
ASTERISK-28951
Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6
This commit adds the endpoint options required to control
Advanced Codec Negotiation.
incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs
The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs. That'll be handled
when things finalize.
This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.
Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11
Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.
This causes unexpected rport handle at the other end.
Added option for disable this behaviour in the pjsip.conf.
This is a system option, but working as a gloabl option.
ASTERISK-28959
Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc
Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is
sent, the caller ID will be checked to see if there is a certificate
that corresponds to it. If so, that information will be retrieved and an
Identity header will be added to the SIP message. The format is:
header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken
Header, payload, and signature are all BASE64 encoded. The public key
URL is retrieved from the certificate. Currently the algorithm and ppt
are ES256 and shaken, respectively. This message is signed and can be
used for verification on the receiving end.
Two new configuration options have been added to the certificate object:
attestation and origid. The attestation is required and must be A, B, or
C. origid is the origination identifier.
A new utility function has been added as well that takes a string,
allocates space, BASE64 encodes it, then returns it, eliminating the
need to calculate the size yourself.
Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4
What's wrong with ast_debug?
ast_debug is fine for general purpose debug output but it's not
really geared for scope tracing since it doesn't present its
output in a way that makes capturing and analyzing flow through
Asterisk easy.
How is scope tracing better?
Scope tracing uses the same "cleanup" attribute that RAII_VAR
uses to print messages to a separate "trace" log level. Even
better, the messages are indented and unindented based on a
thread-local call depth counter. When output to a separate log
file, the output is uncluttered and easy to follow.
Here's an example of the output. The leading timestamps and
thread ids are removed and the output cut off at 68 columns for
commit message restrictions but you get the idea.
--> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
--> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
--> res_pjsip_session.c:3669 handle_incoming_response PJSIP/
--> chan_pjsip.c:3265 chan_pjsip_incoming_response_after
--> chan_pjsip.c:3194 chan_pjsip_incoming_response P
chan_pjsip.c:3245 chan_pjsip_incoming_respon
<-- chan_pjsip.c:3194 chan_pjsip_incoming_response P
<-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after
<-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/
<-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
<-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
The messages with the "-->" or "<--" were produced by including
the following at the top of each function:
SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session));
Scope isn't limited to functions any more than RAII_VAR is. You
can also see entry and exit from "if", "for", "while", etc blocks.
There is also an ast_trace() macro that doesn't track entry or
exit but simply outputs a message to the trace log using the
current indent level. The deepest message in the sample
(chan_pjsip.c:3245) was used to indicate which "case" in a
"select" was executed.
How do you use it?
More documentation is available in logger.h but here's an overview:
* Configure with --enable-dev-mode. Like debug, scope tracing
is #ifdef'd out if devmode isn't enabled.
* Add a SCOPE_TRACE() call to the top of your function.
* Set a logger channel in logger.conf to output the "trace" level.
* Use the CLI (or cli.conf) to set a trace level similar to setting
debug level... CLI> core set trace 2 res_pjsip.so
Summary Of Changes:
* Added LOG_TRACE logger level. Actually it occupies the slot
formerly occupied by the now defunct "event" level.
* Added core asterisk option "trace" similar to debug. Includes
ability to specify global trace level in asterisk.conf and CLI
commands to turn on/off and set levels. Levels can be set
globally (probably not a good idea), or by module/source file.
* Updated sample asterisk.conf and logger.conf. Tracing is
disabled by default in both.
* Added __ast_trace() to logger.c which keeps track of the indent
level using TLS. It's #ifdef'd out if devmode isn't enabled.
* Added ast_trace() and SCOPE_TRACE() macros to logger.h.
These are all #ifdef'd out if devmode isn't enabled.
Why not use gcc's -finstrument-functions capability?
gcc's facility doesn't allow access to local data and doesn't
operate on non-function scopes.
Known Issues:
The only know issue is that we currently don't know the line
number where the scope exited. It's reported as the same place
the scope was entered. There's probably a way to get around it
but it might involve looking at the stack and doing an 'addr2line'
to get the line number. Kind of like ast_backtrace() does.
Not sure if it's worth it.
Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027
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
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
Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.
* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)
* Add "call_direction" to res_pjsip_session.
* Update pjsip_session_caps.c to make the functions more generic
so they could be used for both incoming and outgoing.
* Update ast_sip_session_create_outgoing to create the
pending_media_state->topology with the results of
ast_sip_session_create_joint_call_stream().
* The endpoint "preferred_codec_only" option now automatically sets
AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.
* A helper function ast_stream_get_format_count() was added to
streams to return the current count of formats.
ASTERISK-28777
Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
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.
Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.
This patch does the following:
Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.
Adds a new ast_sip_session_caps structure that's set for each session media
object.
Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.
ASTERISK-28756 #close
Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use. To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes" In Asterisk versions <18, the default
is "no" preserving existing behavior. Beginning with
Asterisk 18, the option will default to "yes".
NOTE: This change does not affect UserEvents or the ARI
TextMessageReceived events.
* Added the "hide_messaging_ami_events" option to asterisk.conf.
* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
the "Message/ast_msg_queue" channel if the option is set in
asterisk.conf. This suppresses the reporting of the events.
Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b
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.
Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660
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
ASTERISK-28710 #close
Change-Id: I647510f796473793b1d3ce1beb32659813be69e1
Adds source port matching support when IP matching is used:
[example]
type = identify
match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444
If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.
ASTERISK-28639 #close
Reported by: Mitch Claborn
Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.
Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa
Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.
Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7
This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.
This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.
ASTERISK-28018
Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06
This patch adds basic Asterisk channel statistics to the res_prometheus
module. This includes:
* asterisk_calls_sum: A running sum of the total number of
processed calls
* asterisk_calls_count: The current number of calls
* asterisk_channels_count: The current number of channels
* asterisk_channels_state: The state of any particular channel
* asterisk_channels_duration_seconds: How long a channel has existed,
in seconds
In all cases, enough information is provided with each channel metric
to determine a unique instance of Asterisk that provided the data, as
well as the name, type, unique ID, and - if present - linked ID of each
channel.
ASTERISK-28403
Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59
Prometheus is the defacto monitoring tool for containerized applications.
This patch adds native support to Asterisk for serving up Prometheus
compatible metrics, such that a Prometheus server can scrape an Asterisk
instance in the same fashion as it does other HTTP services.
The core module in this patch provides an API that future work can build
on top of. The API manages metrics in one of two ways:
(1) Registered metrics. In this particular case, the API assumes that
the metric (either allocated on the stack or on the heap) will have
its value updated by the module registering it at will, and not
just when Prometheus scrapes Asterisk. When a scrape does occur,
the metrics are locked so that the current value can be retrieved.
(2) Scrape callbacks. In this case, the API allows consumers to be
called via a callback function when a Prometheus initiated scrape
occurs. The consumers of the API are responsible for populating
the response to Prometheus themselves, typically using stack
allocated metrics that are then formatted properly into strings
via this module's convenience functions.
These two mechanisms balance the different ways in which information is
generated within Asterisk: some information is generated in a fashion
that makes it appropriate to update the relevant metrics immediately;
some information is better to defer until a Prometheus server asks for
it.
Note that some care has been taken in how metrics are defined to
minimize the impact on performance. Prometheus's metric definition
and its support for nesting metrics based on labels - which are
effectively key/value pairs - can make storage and managing of metrics
somewhat tricky. While a naive approach, where we allow for any number
of labels and perform a lot of heap allocations to manage the information,
would absolutely have worked, this patch instead opts to try to place
as much information in length limited arrays, stack allocations, and
vectors to minimize the performance impacts of scrapes. The author of
this patch has worked on enough systems that were driven to their knees
by poor monitoring implementations to be a bit cautious.
Additionally, this patch only adds support for gauges and counters.
Additional work to add summaries, histograms, and other Prometheus
metric types may add value in the future. This would be of particular
interest if someone wanted to track SIP response types.
Finally, this patch includes unit tests for the core APIs.
ASTERISK-28403
Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address
This causes both 192.168.1.1 and 1.2.3.4 to be advertized.
Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db
When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.
This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.
ASTERISK-28401
Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c
There a long history here:
In commit dd1e62c095 has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf
After that, the commit 4b50e3f1ee fix the
sample file.
We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.
Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c
