Issues:
* The bridging core allowed multiple bridges to be created with the same
unique bridgeId at the same time. Only the last bridge created with the
duplicate name was actually saved to the core bridges container.
* The bridging core was creating a stasis topic for the bridge and saving it
in the bridge->topic field but not increasing its reference count. In the
case where two bridges were created with the same uniqueid (which is also
the topic name), the second bridge would get the _existing_ topic the first
bridge created. When the first bridge was destroyed, it would take the
topic with it so when the second bridge attempted to publish a message to
it it either FRACKed or SEGVd.
* The bridge destructor, which also destroys the bridge topic, is run from the
bridge manager thread not the caller's thread. This makes it possible for
an ARI developer to create a new one with the same uniqueid believing the
old one was destroyed when, in fact, the old one's destructor hadn't
completed. This could cause the new bridge to get the old one's topic just
before the topic was destroyed. When the new bridge attempted to publish
a message on that topic, asterisk could either FRACK or SEGV.
* The ARI bridges resource also allowed multiple bridges to be created with
the same uniqueid but it kept the duplicate bridges in its app_bridges
container. This created a situation where if you added two bridges with
the same "bridge1" uniqueid, all operations on "bridge1" were performed on
the first bridge created and the second was basically orphaned. If you
attempted to delete what you thought was the second bridge, you actually
deleted the first one created.
Changes:
* A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if
a topic already exists for a bridge.
* `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in
resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check
if a bridge with the requested uniqueid already exists and will fail if it
does.
* `bridge_register()` in bridges.c now checks the core bridges container to
make sure a bridge doesn't already exist with the requested uniqueid.
Although most callers of `bridge_register()` will have already called
`bridge_base_init()`, which will now fail on duplicate bridges, there
is no guarantee of this so we must check again.
* The core bridges container allocation was changed to reject duplicate
uniqueids instead of silently replacing an existing one. This is a "belt
and suspenders" check.
* A global mutex was added to bridge.c to prevent concurrent calls to
`bridge_base_init()` and `bridge_register()`.
* Even though you can no longer create multiple bridges with the same uniqueid
at the same time, it's still possible that the bridge topic might be
destroyed while a second bridge with the same uniqueid was trying to use
it. To address this, the bridging core now increments the reference count
on bridge->topic when a bridge is created and decrements it when the
bridge is destroyed.
* `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges
container to make sure a bridge with the requested uniqueid doesn't already
exist. This may seem like overkill but there are so many entrypoints to
bridge creation that we need to be safe and catch issues as soon in the
process as possible.
* The stasis app_bridges container allocation was changed to reject duplicate
uniqueids instead of adding them. This is a "belt and suspenders" check.
* The `bridge show all` CLI command now shows the bridge name as well as the
bridge id.
* Response code 409 "Conflict" was added as a possible response from the ARI
bridge create resources to signal that a bridge with the requested uniqueid
already exists.
* Additional debugging was added to multiple bridging and stasis files.
Resolves: #211
The verification check for missing or anonymous callerid was happening before
the endpoint's profile was retrieved which meant that the failure_action
parameter wasn't available. Therefore, if verification was enabled and there
was no callerid or it was "anonymous", the call was immediately terminated
instead of giving the dialplan the ability to decide what to do with the call.
* The callerid check now happens after the verification context is created and
the endpoint's stir_shaken_profile is available.
* The check now processes the callerid failure just as it does for other
verification failures and respects the failure_action parameter. If set
to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)`
in the dialplan will return "invalid_or_no_callerid".
* If the endpoint's failure_action is "reject_request", the call will be
rejected with `433 "Anonymity Disallowed"`.
* If the endpoint's failure_action is "continue_return_reason", the call will
continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"`
header will be added to the next provisional or final response.
Resolves: #1112
Between ast_ari_channels_external_media(), external_media_rtp_udp(),
and external_media_audiosocket_tcp(), the `variables` structure being passed
around wasn't being cleaned up properly when there was a failure.
* In ast_ari_channels_external_media(), the `variables` structure is now
defined with RAII_VAR to ensure it always gets cleaned up.
* The ast_variables_destroy() call was removed from external_media_rtp_udp().
* The ast_variables_destroy() call was removed from
external_media_audiosocket_tcp(), its `endpoint` allocation was changed to
to use ast_asprintf() as external_media_rtp_udp() does, and it now
returns an error on failure.
* ast_ari_channels_external_media() now checks the new return code from
external_media_audiosocket_tcp() and sets the appropriate error response.
Resolves: #1109
Introduce a ChannelTransfer event and the ability to notify progress to
ARI. Implement emitting this event from the PJSIP channel instead of
handling the transfer in Asterisk when configured.
Introduce a dialplan function to the PJSIP channel to switch between the
"core" and "ari-only" behavior.
UserNote: Call transfers on the PJSIP channel can now be controlled by
ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
dialplan function.
This process was a bit different than the others because everything
is in the same file, there's an array that contains the command
names and their handler functions, and the last command was created
over 15 years ago.
* Dump a `git blame` of res/res_agi.c from BEFORE the handle_* prototypes
were changed.
* Create a command <> handler function xref by parsing the the agi_command
array.
* For each entry, grep the function definition line "static int handle_*"
from the git blame output and capture the commit. This will be the
commit the command was created in.
* Do a `git tag --contains <commit> | sort -V | head -1` to get the
tag the function was created in.
* Add a single since/version element to the command XML. Multiple versions
aren't supported here because the branching and tagging scheme changed
several times in the 2000's.
When an incoming request can't be matched to an endpoint, the "artificial"
auth object is used to create a challenge to return in a 401 response and we
emit a "No matching endpoint found" log message. If the client then responds
with an Authorization header but the request still can't be matched to an
endpoint, the verification will fail and, as before, we'll create a challenge
to return in a 401 response and we emit a "No matching endpoint found" log
message. HOWEVER, because there WAS an Authorization header and it failed
verification, we should have also been emitting a "Failed to authenticate"
log message but weren't because there was a check that short-circuited that
it if the artificial auth was used. Since many admins use the "Failed to
authenticate" message with log parsers like fail2ban, those attempts were not
being recognized as suspicious.
Changes:
* digest_check_auth() now always emits the "Failed to authenticate" log
message if verification of an Authorization header failed even if the
artificial auth was used.
* The verification logic was refactored to be clearer about the handling
of the return codes from verify().
* Comments were added clarify what return codes digest_check_auth() should
return to the distributor and the implications of changing them.
Resolves: #1095
An issue in config_auth.c:ast_sip_auth_digest_algorithms_vector_init() was
causing double allocations for the two supported_algorithms vectors to the
tune of 915 bytes. The leak only happens on startup and when a reload is done
and doesn't get bigger with the number of auth objects defined.
* Pre-initialized the two vectors in config_auth:auth_alloc().
* Removed the allocations in ast_sip_auth_digest_algorithms_vector_init().
* Added a note to the doc for ast_sip_auth_digest_algorithms_vector_init()
noting that the vector passed in should be initialized and empty.
* Simplified the create_artificial_auth() function in pjsip_distributor.
* Set the vector initialization count to 0 in config_global:global_apply().
* Do a git blame on the embedded XML application or function element.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml
* Do a git blame on the embedded XML managerEvent elements.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The events defined in res/res_pjsip/pjsip_manager.xml were done by hand
because the XML was extracted from the C source file relatively recently.
Two bugs were fixed along the way...
* The get_documentation awk script was exiting after it processed the first
DOCUMENTATION block it found in a file. We have at least 1 source file
with multiple DOCUMENTATION blocks so only the first one in them was being
processed. The awk script was changed to continue searching rather
than exiting after the first block.
* Fixing the awk script revealed an issue in logger.c where the third
DOCUMENTATION block contained a XML fragment that consisted only of
a managerEventInstance element that wasn't wrapped in a managerEvent
element. Since logger_doc.xml already existed, the remaining fragments
in logger.c were moved to it and properly organized.
This should resolve the Prometheus error:
> Error scraping target: non-compliant scrape target
sending blank Content-Type and no
fallback_scrape_protocol specified for target.
Resolves: #1075
Most of the configObjects and configOptions that are implemented with
ACO or Sorcery now have `<since>/<version>` elements added. There are
probably some that the script I used didn't catch. The version tags were
determined by the following...
* Do a git blame on the API call that created the object or option.
* From the commit hash, grab the summary line.
* Do a `git log --grep <summary>` to find the cherry-pick commits in all
branches that match.
* Do a `git patch-id` to ensure the commits are all related and didn't get
a false match on the summary.
* Do a `git tag --contains <commit>` to find the tags that contain each
commit.
* Weed out all tags not <major>.<minor>.0.
* Sort and discard any <major>.0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the API was last touched.
configObjects and configOptions elements implemented with the base
ast_config APIs were just not possible to find due to the non-deterministic
way they are accessed.
Also note that if the API call was on modified after it was added, the
version will be the one it was last modified in.
Final note: The configObject and configOption elements were introduced in
12.0.0 so options created before then may not have any XML documentation.
The return code fom digest_check_auth wasn't explicitly being initialized.
The return code also wasn't explicitly set to CHALLENGE when challenges
were sent. When optimization was turned off (DONT_OPTIMIZE), the compiler
was setting it to "0"(CHALLENGE) which worked fine. However, with
optimization turned on, it was setting it to "1" (SUCCESS) so if there was
no incoming Authorization header, the function was returning SUCCESS to the
distributor allowing the request to incorrectly succeed.
The return code is now initialized correctly and is now explicitly set
to CHALLENGE when we send challenges.
* channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element.
* main/bucket.c: Removed XML completely since the "bucket" and "file" objects
are internal only with no config file.
* main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf"
and should have been "acl.conf"
* res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element.
* res/res_http_media_cache.c: Fixed the configFile element name. It was
"http_media_cache.conf" and should have been "res_http_media_cache.conf".
* Added the "since" element to the XML configObject and configOption elements
in appdocsxml.dtd.
* Added the "Since" section to the following CLI output:
```
config show help <module> <object>
config show help <module> <object> <option>
core show application <app>
core show function <func>
manager show command <command>
manager show event <event>
agi show commands topic <topic>
```
* Refactored the commands above to output their sections in the same order:
Synopsis, Since, Description, Syntax, Arguments, SeeAlso
* Refactored the commands above so they all use the same pattern for writing
the output to the CLI.
* Fixed several memory leaks caused by failure to free temporary output
buffers.
* Added a "since" array to the mustache template for the top-level resources
(Channel, Endpoint, etc.) and to the paths/methods underneath them. These
will be added to the generated markdown if present.
Example:
```
"resourcePath": "/api-docs/channels.{format}",
"requiresModules": [
"res_stasis_answer",
"res_stasis_playback",
"res_stasis_recording",
"res_stasis_snoop"
],
"since": [
"18.0.0",
"21.0.0"
],
"apis": [
{
"path": "/channels",
"description": "Active channels",
"operations": [
{
"httpMethod": "GET",
"since": [
"18.6.0",
"21.8.0"
],
"summary": "List all active channels in Asterisk.",
"nickname": "list",
"responseClass": "List[Channel]"
},
```
NOTE: No versioning information is actually added in this commit.
Those will be added separately and instructions for adding and maintaining
them will be published on the documentation site at a later date.
* Refactored pjproject code to support the new algorithms and
added a patch file to third-party/pjproject/patches
* Added new parameters to the pjsip auth object:
* password_digest = <algorithm>:<digest>
* supported_algorithms_uac = List of algorithms to support
when acting as a UAC.
* supported_algorithms_uas = List of algorithms to support
when acting as a UAS.
See the auth object in pjsip.conf.sample for detailed info.
* Updated both res_pjsip_authenticator_digest.c (for UAS) and
res_pjsip_outbound_authentocator_digest.c (UAC) to suport the
new algorithms.
The new algorithms are only available with the bundled version
of pjproject, or an external version > 2.14.1. OpenSSL version
1.1.1 or greater is required to support SHA-512-256.
Resolves: #948
UserNote: The SHA-256 and SHA-512-256 algorithms are now available
for authentication as both a UAS and a UAC.
Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify
feature to mark a contact as available only if an OPTIONS request
returns a 2XX response. If the option is not specified or is false,
any response to the OPTIONS request marks the contact as available.
UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only"
option that can be set so that only 2XX responses to OPTIONS requests
used to qualify a contact will mark the contact as available.
Whenever a slot is freed up due to a failed connection, wake up a waiter
before failing.
In the case of a dead connection there could be waiters, for example,
let's say two threads tries to acquire objects at the same time, with
one in the cached connections, one will acquire the dead connection, and
the other will enter into the wait state. The thread with the dead
connection will clear up the dead connection, and then attempt a
re-acquire (at this point there cannot be cached connections else the
other thread would have received that and tried to clean up), as such,
at this point we're guaranteed that either there are no waiting threads,
or that the maxconnections - connection_cnt threads will attempt to
re-acquire connections, and then either succeed, using those
connections, or failing, and then signalling to release more waiters.
Also fix the pointer log for ODBC handle %p dead which would always
reflect NULL.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Added a new option "unknown_tn_attest_level" to allow Identity
headers to be sent when a callerid TN isn't explicitly configured
in stir_shaken.conf. Since there's no TN object, a private_key_file
and public_cert_url must be configured in the attestation or profile
objects.
Since "unknown_tn_attest_level" uses the same enum as attest_level,
some of the sorcery macros had to be refactored to allow sharing
the enum and to/from string conversion functions.
Also fixed a memory leak in crypto_utils:pem_file_cb().
Resolves: #921
UserNote: You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.
The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.
Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.
Resolves: #995
Normally, when one party in a call sends Asterisk an SDP with
a "sendonly" or "inactive" attribute it means "hold" and causes
Asterisk to start playing MOH back to the other party. This can be
problematic if it happens at certain times, such as in a 183
Progress message, because the MOH will replace any early media you
may be playing to the calling party. If you set this option
to "yes" on an endpoint and the endpoint receives an SDP
with "sendonly" or "inactive", Asterisk will NOT play MOH back to
the other party.
Resolves: #979
UserNote: The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
The tenantid field was originally added to the ast_sip_endpoint
structure at the end of the AST_DECLARE_STRING_FIELDS block. This
caused everything after it in the structure to move down in memory
and break ABI compatibility. It's now at the end of the structure
as an AST_STRING_FIELD_EXTENDED. Given the number of string fields
in the structure now, the initial string field allocation was
also increased from 64 to 128 bytes.
Resolves: #982
The key used for transport monitors was the remote host name for the
transport and not the remote address resolved for this domain.
This was problematic for domains returning multiple addresses as several
transport monitors were created with the same key.
Whenever a subsystem wanted to register a callback it would always end
up attached to the first transport monitor with a matching key.
The key used for transport monitors is now the remote address and port
the transport actually connected to.
Fixes: #932
There's really no point in spamming logs with a verbose message
for every unsupported crypto suite an older client may send
in an SDP. If none are supported, there will be an error or
warning.
When a transport is disconnected, several events can arrive following
each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it
will trigger the destruction of the transport monitor object. The lookup
for the transport monitor to destroy is done using the transport key,
that contains the transport destination host:port.
A reconnect attempt by pjsip will be triggered as soon something needs to
send a packet using that transport. This can happen directly after a
disconnect since ca
Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will
also try to trigger the destruction of the transport monitor if not
already done. Since the lookup for the transport monitor to destroy is
done using the transport key, it can match newly created transports
towards the same destination and destroy their monitor object.
Because of this, it was sometimes not possible to monitor a transport
after one or more disconnections.
This fix adds an additional check on the transport pointer to ensure
only a monitor for that specific transport is removed.
Fixes: #923
This reverts commit cb5e3445be.
The original change from 16 to 15 bit sequence numbers was predicated
on the following from the now-defunct libSRTP FAQ on sourceforge.net:
> *Q6. The use of implicit synchronization via ROC seems
> dangerous. Can senders and receivers lose ROC synchronization?*
>
> **A.** It is possible to lose ROC synchronization between sender and
> receiver(s), though it is not likely in practice, and practical
> steps can be taken to avoid it. A burst loss of 2^16 packets or more
> will always break synchronization. For example, a conversational
> voice codec that sends 50 packets per second will have its ROC
> increment about every 22 minutes. A network with a burst of packet
> loss that long has problems other than ROC synchronization.
>
> There is a higher sensitivity to loss at the very outset of an SRTP
> stream. If the sender's initial sequence number is close to the
> maximum value of 2^16-1, and all packets are lost from the initial
> packet until the sequence number cycles back to zero, the sender
> will increment its ROC, but the receiver will not. The receiver
> cannot determine that the initial packets were lost and that
> sequence-number rollover has occurred. In this case, the receiver's
> ROC would be zero whereas the sender's ROC would be one, while their
> sequence numbers would be so close that the ROC-guessing algorithm
> could not detect this fact.
>
> There is a simple solution to this problem: the SRTP sender should
> randomly select an initial sequence number that is always less than
> 2^15. This ensures correct SRTP operation so long as fewer than 2^15
> initial packets are lost in succession, which is within the maximum
> tolerance of SRTP packet-index determination (see Appendix A and
> page 14, first paragraph of RFC 3711). An SRTP receiver should
> carefully implement the index-guessing algorithm. A naive
> implementation can unintentionally guess the value of
> 0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15
> and the locally stored SEQ and ROC are zero. (This can happen when
> the implementation fails to treat those zero values as a special
> case.)
>
> When ROC synchronization is lost, the receiver will not be able to
> properly process the packets. If anti-replay protection is turned
> on, then the desynchronization will appear as a burst of replay
> check failures. Otherwise, if authentication is being checked, then
> it will appear as a burst of authentication failures. Otherwise, if
> encryption is being used, the desynchronization may not be detected
> by the SRTP layer, and the packets may be improperly decrypted.
However, modern libSRTP (as of 1.0.1[1]) now mentions the following in
their README.md[2]:
> The sequence number in the rtp packet is used as the low 16 bits of
> the sender's local packet index. Note that RTP will start its
> sequence number in a random place, and the SRTP layer just jumps
> forward to that number at its first invocation. An earlier version
> of this library used initial sequence numbers that are less than
> 32,768; this trick is no longer required as the
> rdbx_estimate_index(...) function has been made smarter.
So truncating our initial sequence number to 15 bit is no longer
necessary.
1. 0eb007f0dc/CHANGES (L271-L289)
2. 2de20dd9e9/README.md (implementation-notes)
Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be
balanced to ensure that we can capture the exit status of child
processes when we need to. This extends to functions that call
`ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as
`ast_safe_fork()` and `ast_safe_fork_cleanup()`.
The primary change here is ensuring that we do not call
`ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously
called `ast_safe_fork()`.
Additionally we reinforce some of the documentation and add an
assertion to, ideally, catch this sooner were this to happen again.
Fixes#922
asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings.
config.c: Include header file for WIFEXITED/WEXITSTATUS macros.
res_timing_kqueue: Use more portable format specifier.
test_crypto: Use non-linux limits.h header file.
Resolves: #916
In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned
success but with a timeout of 0, we were stopping the timer and
decrementing the refcount on instance but not resetting the
timeout_timer to -1. When dtls_srtp_stop_timeout_timer()
was later called, it was atempting to stop a stale timer and could
decrement the refcount on instance again which would then cause
the instance destructor to run early. This would result in either
a FRACK or a SEGV when ast_rtp_stop(0 was called.
According to the OpenSSL docs, we shouldn't have been stopping the
timer when DTLSv1_get_timeout() returned success and the new timeout
was 0 anyway. We should have been calling DTLSv1_handle_timeout()
again immediately so we now reschedule the timer callback for
1ms (almost immediately).
Additionally, instead of scheduling the timer callback at a fixed
interval returned by the initial call to DTLSv1_get_timeout()
(usually 999 ms), we now reschedule the next callback based on
the last call to DTLSv1_get_timeout().
Resolves: #487
When using the speech recognition module, crashes can occur
sporadically due to a "double free or corruption (out)" error. Now, in
the section where the audio stream is being captured in a loop, each
time after releasing fr, it is set to NULL to prevent repeated
deallocation.
Fixes#772
attest_level, send_mky and check_tn_cert_public_url weren't
propagating correctly from the attestation object to the profile
and tn.
* In the case of attest_level, the enum needed to be changed
so the "0" value (the default) was "NOT_SET" instead of "A". This
now allows the merging of the attestation object, profile and tn
to detect when a value isn't set and use the higher level value.
* For send_mky and check_tn_cert_public_url, the tn default was
forced to "NO" which always overrode the profile and attestation
objects. Their defaults are now "NOT_SET" so the propagation
happens correctly.
* Just to remove some redundant code in tn_config.c, a bunch of calls to
generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
replaced with a single call to generate_acfg_common_sorcery_handlers().
Resolves: #904
verification.c had an include for jansson.h left over from previous
versions of the module. Since res_stir_shaken no longer has a
dependency on jansson, the bundled version wasn't added to GCC's
include path so if you didn't also have a jansson development package
installed, the compile would fail. Removing the stale include
was the only thing needed.
Resolves: #889
* If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
check_for_old_config() now returns LOAD_DECLINE instead of continuing
on with a bad pointer.
* If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
assumes the config is being loaded from realtime and now returns
LOAD_SUCCESS. If it's actually not being loaded from realtime,
sorcery will catch that later on.
* Also refactored the error handling in load_module() a bit.
Resolves: #884
For both attestation and verification, we now check whether they've
been disabled either globally or by the profile before validating
things like callerid, orig_tn, dest_tn, etc. This prevents useless
error messages.
Resolves: #879
The ub_result pointer passed to unbound_resolver_callback by
libunbound can be NULL if the query was for something malformed
like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result
and return instead of crashing with a SEGV. This causes pjproject
to simply cancel the transaction with a "No answer record in the DNS
response" error. The existing "off nominal" unit test was also
updated to check this condition.
Although not necessary for this fix, we also made
ast_dns_resolver_completed() tolerant of a NULL result.
Resolves: GHSA-v428-g3cw-7hv9
When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
matching 4733 offers, Bob may want to use the 48K audio codec but can not
accept 48K digits and so negotiates for a mixed set.
Asterisk will now check Bob's offer to make sure Bob has indicated this is
acceptible and if not, will use Bob's preference.
Fixes: #847
* A static array of security mechanism type names was created.
* ast_sip_str_to_security_mechanism_type() was refactored to do
a lookup in the new array instead of using fixed "if/else if"
statments.
* security_mechanism_to_str() and ast_sip_security_mechanisms_to_str()
were refactored to use ast_str instead of a fixed length buffer
to store the result.
* ast_sip_security_mechanism_type_to_str was removed in favor of
just referencing the new type name array. Despite starting with
"ast_sip_", it was a static function so removing it doesn't affect
ABI.
* Speaking of "ast_sip_", several other static functions that
started with "ast_sip_" were renamed to avoid confusion about
their public availability.
* A few VECTOR free loops were replaced with AST_VECTOR_RESET().
* Fixed a meomry leak in pjsip_configuration.c endpoint_destructor
caused by not calling ast_sip_security_mechanisms_vector_destroy().
* Fixed a memory leak in res_pjsip_outbound_registration.c
add_security_headers() caused by not specifying OBJ_NODATA in
an ao2_callback.
* Fixed a few ao2_callback return code misuses.
Resolves: #845
PR #700 added a preferred_format for the struct ast_rtp_codecs,
but when set the preferred_format it leaks an astobj2 ast_format.
In the next code
ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
both functions ast_rtp_codecs_set_preferred_format
and ast_format_cap_get_format increases the ao2 reference count.
Fixes: #856
Add dialplan application PJSIPNOTIFY to send either pre-configured
NOTIFY messages from pjsip_notify.conf or with headers defined in
dialplan.
Also adds the ability to send pre-configured NOTIFY commands to a
channel via the CLI.
Resolves: #799
UserNote: A new dialplan application PJSIPNotify is now available
which can send SIP NOTIFY requests from the dialplan.
The pjsip send notify CLI command has also been enhanced to allow
sending NOTIFY messages to a specific channel. Syntax:
pjsip send notify <option> channel <channel>
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
A stasis event is now produced when using the TONE_DETECT dialplan
function. This event is published over ARI using the ChannelToneDetected
event. This change does not make it available over AMI.
Fixes: #811
UserNote: Setting the TONE_DETECT dialplan function on a channel
in ARI will now cause a ChannelToneDetected ARI event to be raised
when the specified tone is detected.
Previously, on command execution, the control thread was awoken by
sending a SIGURG. It was found that this still resulted in some
instances where the thread was not immediately awoken.
This change instead sends a null frame to awaken the control thread,
which awakens the thread more consistently.
Resolves: #801
When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.
Resolves: asterisk#826
* Fixed a bug in crypto_show_cli_store that was causing asterisk
to crash if there were certificate revocation lists in the
verification certificate store. We're also now prefixing
certificates with "Cert:" and CRLs with "CRL:" to distinguish them
in the list.
* Added 'untrusted_cert_file' and 'untrusted_cert_path' options
to both verification and profile objects. If you have CRLs that
are signed by a different CA than the incoming X5U certificate
(indirect CRL), you'll need to provide the certificate of the
CRL signer here. Thse will show up as 'Untrusted" when showing
the verification or profile objects.
* Fixed loading of crl_path. The OpenSSL API we were using to
load CRLs won't actually load them from a directory, only a file.
We now scan the directory ourselves and load the files one-by-one.
* Fixed the verification flags being set on the certificate store.
- Removed the CRL_CHECK_ALL flag as this was causing all certificates
to be checked for CRL extensions and failing to verify the cert if
there was none. This basically caused all certs to fail when a CRL
was provided via crl_file or crl_path.
- Added the EXTENDED_CRL_SUPPORT flag as it is required to handle
indirect CRLs.
* Added a new CLI command...
`stir_shaken verify certificate_file <certificate_file> [ <profile> ]`
which will assist troubleshooting certificate problems by allowing
the user to manually verify a certificate file against either the
global verification certificate store or the store for a specific
profile.
* Updated the XML documentation and the sample config file.
Resolves: #809
The way we have been initializing the config wizard prevented it
from registering its objects if res_pjsip happened to load
before it.
* We now use the object_type_registered sorcery observer to kick
things off instead of the wizard_mapped observer.
* The load_module function now checks if res_pjsip has been loaded
already and if it was it fires the proper observers so the objects
load correctly.
Resolves: #816
UserNote: The res_pjsip_config_wizard.so module can now be reloaded.
When using the PJSIP_DIAL_CONTACTS() function for use in the Dial()
command, the contacts are returned in text form, so the input to
the path_outgoing_request() function is a contact value of NULL.
The issue was reported in ASTERISK-28211, but was not actually fixed
in ASTERISK-30100. This fix brings back the code that was previously
removed and adds code to search for a contact to extract the path
value from it.
After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
the negotiated codec(s).
This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.
A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
be re-written to allow for these scenarios.
Fixes: #776
The `Require: mediasec` and `Proxy-Require: mediasec` headers need
to be sent whenever we send `Security-Client` or `Security-Verify`
headers but the logic to do that was only in add_security_headers()
in res_pjsip_outbound_register. So while we were sending them on
REGISTER requests, we weren't sending them on INVITE requests.
This commit moves the logic to send the two headers out of
res_pjsip_outbound_register:add_security_headers() and into
security_agreement:ast_sip_add_security_headers(). This way
they're always sent when we send `Security-Client` or
`Security-Verify`.
Resolves: #789
Two functions are deprecated as of libxml2 2.12:
* xmlSubstituteEntitiesDefault
* xmlParseMemory
So we update those with supported API.
Additionally, `res_calendar_caldav` has been updated to use libxml2's
xmlreader API instead of the SAX2 API which has always felt a little
hacky (see deleted comment block in `res_calendar_caldav.c`).
The xmlreader API has been around since libxml2 2.5.0 which was
released in 2003.
Fixes#725
Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
Asterisk currently treats RFC2833 Digits as a single rtp payload type
with a fixed bitrate of 8K. This change would expand that to 8, 16,
24 and 32K.
This requires checking the offered rtp types for any of these bitrates
and then adding an offer for each (if configured for RFC2833.) DTMF
generation must also be changed in order to look at the current outbound
codec in order to generate appropriately timed rtp.
For cases where no outgoing audio has yet been sent prior to digit
generation, Asterisk now has a concept of a 'preferred' codec based on
offer order.
On inbound calls Asterisk will mimic the payload types of the RFC2833
digits.
On outbound calls Asterisk will choose the next free payload types starting
with 101.
UserNote: No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
Include signal.h to avoid the following build failure with uclibc-ng
raised since
2694792e13:
stasis/control.c: In function 'exec_command_on_condition':
stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration]
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~~~~~~~
| pthread_yield
stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function)
313 | pthread_kill(control->control_thread, SIGURG);
| ^~~~~~
cherry-pick-to: 18
cherry-pick-to: 20
cherry-pick-to: 21
Fixes: #729
Currently, reloading res_pjsip will cause logging
to be disabled. This is because logging can also
be controlled via the debug option in pjsip.conf
and this defaults to "no".
To improve this, logging is no longer disabled on
reloads if logging had not been previously
enabled using the debug option from the config.
This ensures that logging enabled from the CLI
will persist through a reload.
ASTERISK-29912 #close
Resolves: #246
UserNote: Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.
Fixes#710
* Fixed possible memory leak in tn_config:tn_get_etn() where we
weren't releasing etn if tn or eprofile were null.
* We now canonicalize TNs before using them for lookups or adding
them to Identity headers.
* Fixed a typo in stir_shaken.conf.sample.
Resolves: #716
Add a new identify_by option to res_pjsip_endpoint_identifier_ip
called 'transport' this matches endpoints based on the bound
ip address (local) instead of the 'ip' option, which matches on
the source ip address (remote).
UserNote: set identify_by=transport for the pjsip endpoint. Then
use the existing 'match' option and the new 'transport' option of
the identify.
Fixes: #672
* OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use
X509_get_pubkey. The difference is that X509_get_pubkey requires
the caller to free the EVP_PKEY themselves so we now let
RAII_VAR do that.
* OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now
wrap it in an ao2 object.
* OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all
the certs from an X509_STORE and there's no easy way to polyfill
it so the CLI commands that list profiles will show a "not
supported" message instead of listing the certs in a store.
Resolves: #676
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
Add ability to match against PJSIP request URI.
UserNote: this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary.
Fixes#657
There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip.
Fixes: #579
UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
The prometheus exposition format requires each line to be unique[1].
This is handled by struct prometheus_metric having a list of children
that is managed when registering a metric. In case the scrape callback
is used, it is the responsibility of the implementation to handle this
correctly.
Originally the bridge callback didn't handle NULL snapshots, the crash
fix lead to NULL metrics, and fixing that lead to duplicates.
The original code assumed that snapshots are not NULL and then relied on
"if (i > 0)" to establish the parent/children relationship between
metrics of the same class. This is not workerable as the first bridge
might be invisible/lacks a snapshot.
Fix this by keeping a separate array of the first metric by class.
Instead of relying on the index of the bridge, check whether the array
has an entry. Use that array for the output.
Add a test case that verifies that the help text is not duplicated.
Resolves: #642
[1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting
Currently, if a parking lot is full, bridge setup returns -1,
causing dialplan execution to terminate without TryExec.
However, such failures should be handled more gracefully,
the same way they are on other paths, as indicated by the
module's author, here:
http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html
Now, callers will hear the parking failure announcement, and dialplan
will continue, which is consistent with existing failure modes.
Resolves: #624
In handle_negotiated_sdp the pending_media_state->read_callbacks must be
reset before they are added in the SDP handlers in
handle_negotiated_sdp_session_media. Otherwise, old callbacks for
removed streams and file descriptors could be added to the channel and
Asterisk would poll on non-existing file descriptors.
Resolves: #611
* Added checks for missing session, session->channel and rdata
in stir_shaken_incoming_request.
* Added checks for missing session, session->channel and tdata
in stir_shaken_outgoing_request.
Resolves: #645
In as_check_common_config, we were calling ast_std_free on
raw_key but raw_key was allocated with ast_malloc so it
should be freed with ast_free.
Resolves: #636
Why do we need a refactor?
The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation. The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.
There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.
Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use. With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.
What's changed?
* Configuration objects have been refactored to be clearer about
their uses and to fix issues.
* The "general" object was renamed to "verification" since it
contains parameters specific to the incoming verification
process. It also never handled ca_path and crl_path
correctly.
* A new "attestation" object was added that controls the
outgoing attestation process. It sets default certificates,
keys, etc.
* The "certificate" object was renamed to "tn" and had it's key
change to telephone number since outgoing call attestation
needs to look up certificates by telephone number.
* The "profile" object had more parameters added to it that can
override default parameters specified in the "attestation"
and "verification" objects.
* The "store" object was removed altogther as it was never
implemented.
* We now use libjwt to create outgoing Identity headers and to
parse and validate signatures on incoming Identiy headers. Our
previous custom implementation was much of the source of the
interoperability issues.
* General code cleanup and refactor.
* Moved things to better places.
* Separated some of the complex functions to smaller ones.
* Using context objects rather than passing tons of parameters
in function calls.
* Removed some complexity and unneeded encapsuation from the
config objects.
Resolves: #351Resolves: #46
UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
Media Experience Score relies on incorrect pseudo_mos variable
calculation. According to forming an opinion section of the
documentation, calculation relies on ITU-T G.107 standard:
https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion
ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
"seven times ten to the power of negative six", 7 * 10^(-6). which
would mean 6 digits after the decimal point. Current implementation
has 7 digits after the decimal point, which downrates the calls.
Fixes: #597
When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Resolves: #582
This introduces a setting for outbound registrations to override the
global User-Agent header setting.
Resolves: #515
UserNote: PJSIP outbound registrations now support a per-registration
User-Agent header
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.
Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:
```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```
And that value would be blindly copied into the outgoing SDP without
regard to its format.
Fixes#541
This reverts commit 315eb551db.
Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests. This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages. It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.
Resolves: #530
* Since ICE candidates are used for the check and pjproject is
required to use ICE, res_rtp_asterisk was failing to compile
when pjproject wasn't available. The check is now wrapped
with an #ifdef HAVE_PJPROJECT.
* The rtp->ice_active_remote_candidates container was being
used to check the address on incoming packets but that
container doesn't contain peer reflexive candidates discovered
during negotiation. This was causing the check to fail
where it shouldn't. We now check against pjproject's
real_ice->rcand array which will contain those candidates.
* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
we weren't zeroing out sin->sin_zero before returning. This
was causing ast_sockaddr_cmp() to always return false when
one of the inputs was converted from a pj_sockaddr, even
if both inputs had the same address and port.
Resolves: #500Resolves: #503Resolves: #505
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.
This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
Add patch to split the log level for invalid packets received on the
signaling port. The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.
Resolves: GHSA-hxj9-xwr8-w8pq
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36.
Resolves: #430
There are valid scenarios where res_odbc's connection pool might have some dead
or stuck connections while others are healthy (imagine network
elements/firewalls/routers silently timing out connections to a single DB and a
single IP address, or a heterogeneous connection pool connected to potentially
multiple IPs/instances of a replicated DB using a DNS front end for load
balancing and one replica fails).
In order to time out those unhealthy connections without blocking access to
other parts of Asterisk that may attempt access to the connection pool, it would
be beneficial to not lock/block access around the entire pool in
_ast_odbc_request_obj2 while doing potentially blocking operations on connection
pool objects such as the connection_dead() test, odbc_obj_connect(), or by
dereferencing a struct odbc_obj for the last time and triggering a
odbc_obj_disconnect().
This would facilitate much quicker and concurrent timeout of dead connections
via the connection_dead() test, which could block potentially for a long period
of time depending on odbc.ini or other odbc connector specific timeout settings.
This also would make rapid failover (in the clustered DB scenario) much quicker.
This patch changes the locking in _ast_odbc_request_obj2() to not lock around
odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
lock around truly shared, non-immutable state like the connection_cnt member and
the connections list on struct odbc_class.
Fixes: #465
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
George Joseph
Holger Hans Peter Freyther
Sean Bright
Artem Umerov
Abdelkader Boudih
Stanislav Abramenkov
Kent
Jaco Kroon
Viktor Litvinov
Alexey Vasilyev
Thomas Guebels
Ben Ford
Naveen Albert
jiangxc
Mike Bradeen
Alexei Gradinari
gibbz00
Tinet-mucw
Igor Goncharovsky
Fabrice Fontaine
Henrik Liljedahl
Sperl Viktor
Joshua Elson
Martin Tomec
Martin Nystroem
Maximilian Fridrich
romryz
Flole998
Gitea
Matthew Fredrickson
Brad Smith