In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.
ASTERISK-29402
Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557
The behavior of max_contacts and remove_existing are connected. If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact. Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.
This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing. If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.
ASTERISK-29525
Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784
Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.
ASTERISK-29529
Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702
This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.
ASTERISK-29508 #close
Change-Id: I7955a046293f913ba121bbd82153b04439e3465f
This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.
In particular the behaviours that this impacts include:
- FGC-CAMA did not work at all when using MF signaling. Modified the
switch case block to send calls to the correct part of the
signaling-handling state machine.
- For FGC-CAMA operation, the delay between called number ST and
second wink for ANI spill has been made configurable; previously
all calls were made to wait for one full second.
- After the ANI spill, previous behavior was to require a 'ST' tone
to advance the call. This has been changed to allow 'STP' 'ST2P'
or 'ST3P' as well, for compatibility with ANI-D.
- Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.
- For calls with an ANI failure, No. 1 Crossbar switches will send
forward a single-digit failure code, with no calling number digits
and no ST pulse to terminate the spill. I've made the ANI timeout
configurable so to reduce dead air time on calls with ANI fail.
- ANI info digits configurable. Modern digital switches will send 2
digits, but ANI-B sends only a single info digit. This caused the
ANI reported by Asterisk to be misaligned.
- Changed a confusing log message to be more informative.
ASTERISK-29518
Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256
Allow mapping pjproject log messages to the Asterisk TRACE
log level. The defaults were also changes to log pjproject
levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6
all went to DEBUG.
ASTERISK-29582
Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d
Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.
A flag has been introduced to allow meters to fallback to counters.
ASTERISK-29513
Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7
A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.
ASTERISK-29440
Change-Id: I26642729d0345f178c7b8045506605c8402de54b
By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:
https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2
This patch adds a `stun_software_attribute` yes/no option to make it
possible to omit the SOFTWARE attribute from STUN packets.
ASTERISK-29434
Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b
RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm. The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred. We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.
The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.
It doesn't stop there though... Each realm can require a
different username from the others. There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.
So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header. We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.
In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.
NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5. We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5. When we support the
others, we'll move the check into the loop that searches the
objects.
Changes:
* Added a new API ast_sip_retrieve_auths_vector() that takes in
a vector of auth ids (usually supplied on a call to
ast_sip_create_request_with_auth()) and populates another
vector with the actual objects.
* Refactored res_pjsip_outbound_authenticator_digest to handle
multiple Authenticate headers and set the stage for handling
additional digest algorithms.
* Added a pjproject patch that allows them to ignore digest
algorithms they don't support. This patch has already been
merged upstream.
* Updated documentation for auth objects in the XML and
in pjsip.conf.sample.
* Although res_pjsip_authenticator_digest isn't affected
by this change, some debugging and a testsuite AMI event
was added to facilitate testing.
Discovered during OpenSIPit 2021.
ASTERISK-29397
Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
ASTERISK-29380
Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81
STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.
The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c
During OpenSIPit, we found out that the public certificates must be of
type X.509. When reading in public keys, we use the corresponding X.509
functions now.
We also discovered that we needed a better naming scheme for the
certificates since certificates with the same name would cause issues
(overwriting certs, etc.). Now when we download a public certificate, we
get the serial number from it and use that as the name of the cached
certificate.
The configuration option public_key_url in stir_shaken.conf has also
been renamed to public_cert_url, which better describes what the option
is for.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d
Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.
ASTERISK-26614 #close
Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217
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, but leaves
it at /tmp by default, to ensure backwards compatibility.
A future commit that only targets master could change the default
location to something more sensible such as /var/tmp/asterisk. At
that point, the cachedir could be created and cleaned up during
uninstall by the Makefile script.
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
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
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
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
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
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
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
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
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>
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
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
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
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
