* 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
(cherry picked from commit d010dd9752)
ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.
Resolves: #876
(cherry picked from commit 03983b2ff9)
It's possible that ast_autoservice_stop is called within the autoservice thread.
In this case the autoservice thread is stuck in an endless sleep.
To avoid endless sleep ast_autoservice_stop must check that it's not called
within the autoservice thread.
Fixes: #763
(cherry picked from commit 9b3d3a7f0e)
When commit e8c9cb80 was cherry-picked in from master, the
fact that the 20 and 18 branches still had the old "macrocontext"
column wasn't taken into account so the number of named parameters
didn't match the number of '?' placeholders. They do now.
We also now use ast_asprintf to create the full mailbox query SQL
statement instead of trying to calculate the proper length ourselves.
Resolves: #831
(cherry picked from commit 83b3783499)
When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
matching 4733 offers, Bob may want to use the 48K audio codec but can not
accept 48K digits and so negotiates for a mixed set.
Asterisk will now check Bob's offer to make sure Bob has indicated this is
acceptible and if not, will use Bob's preference.
Fixes: #847
(cherry picked from commit cf5a6435c2)
ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.
Resolves: #861
(cherry picked from commit 4263f7b58d)
* 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
(cherry picked from commit 53c76478f2)
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
(cherry picked from commit 8b39a956e7)
On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)
(cherry picked from commit 5840e55bb5)
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>
(cherry picked from commit e94c5f0d3b)
If you run an AMI CoreShowChannelMap on a channel that isn't in a
bridge and you're in DEVMODE, you can get a FRACK because the
bridge id is empty. We now simply return an empty list for that
request.
(cherry picked from commit 2da5559482)
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.
(cherry picked from commit 027127246e)
source is a bash concept, so when /bin/sh points to another shell the
existing construct won't work.
Reference: https://bugs.gentoo.org/927055
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 7fb7e2b9fd)
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
(cherry picked from commit 25c38db9a4)
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
(cherry picked from commit 4aae8194b4)
* 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
(cherry picked from commit 96cf337cc9)
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.
(cherry picked from commit c9e23c46b0)
softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else()
with the bridge_channel so the VIDUPDATE control frame isn't echoed back.
softmix_bridge_write_control() was setting bridge_channel to NULL
when calling ast_bridge_queue_everyone_else() for VIDUPDATE control
frames. This was causing the frame to be echoed back to the
channel it came from. In certain cases, like when two channels or
bridges are being recorded, this can cause a ping-pong effect that
floods the system with VIDUPDATE control frames.
Resolves: #780
(cherry picked from commit 591c1c77c7)
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
Added Reload and DBdeltree to the list of dialplan application that
can't be executed via the Originate manager action without also
having write SYSTEM permissions.
Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan
functions that can't be executed via the Originate manager action
without also having write SYSTEM permissions.
If the Queue application is attempted to be run by the Originate
manager action and an AGI parameter is specified in the app data,
it'll be rejected unless the manager user has either the AGI or
SYSTEM permissions.
Resolves: #GHSA-c4cg-9275-6w44
There can be empty slots in payload_mapping_tx corresponding to
dynamic payload types that haven't been seen before so we now
check for NULL before attempting to use 'type' in the call to
ast_format_cmp.
Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
Resolves: #822
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.
(cherry picked from commit caa1820d2b)
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
(cherry picked from commit ac9c510d99)
Remove duplicate creation of ast_bool_values from
2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was
causing alembic upgrades to fail since the enum was already created
in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.
Resolves: #797
(cherry picked from commit 614d71fa19)
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
(cherry picked from commit 1d9f43b5a5)
Whenver a new channel snapshot is created or when a channel is
destroyed, we need to delete any existing channel snapshot from
the snapshot cache. Historically, we used the channel->snapshot
pointer to delete any existing snapshots but this has two issues.
First, if something (possibly ast_channel_internal_swap_snapshots)
sets channel->snapshot to NULL while there's still a snapshot in
the cache, we wouldn't be able to delete it and it would be orphaned
when the channel is destroyed. Since we use the cache to list
channels from the CLI, AMI and ARI, it would appear as though the
channel was still there when it wasn't.
Second, since there are actually two caches, one indexed by the
channel's uniqueid, and another indexed by the channel's name,
deleting from the caches by pointer requires a sequential search of
all of the hash table buckets in BOTH caches to find the matching
snapshots. Not very efficient.
So, we now delete from the caches using the channel's uniqueid
and name. This solves both issues.
This doesn't address how channel->snapshot might have been set
to NULL in the first place because although we have concrete
evidence that it's happening, we haven't been able to reproduce it.
Resolves: #783
(cherry picked from commit 6f67835b4f)
This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
which when set causes the ODBC variant of app_voicemail to leave
the message and greeting audio files on disk and only store the
message metadata in the database. This option came from a concern
that the database could grow to large and cause remote access
and/or replication to become slow. In a clustering situation
with this option, all asterisk instances would share the same
database for the metadata and either use a shared filesystem
or other filesystem replication service much more suitable
for synchronizing files.
The changes to app_voicemail to implement this feature were actually
quite small but due to the complexity of the module, the actual
source code changes were greater. They fall into the following
categories:
* Tracing. The module is so complex that it was impossible to
figure out the path taken for various scenarios without the addition
of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
code that's not related to the functional change. Making this worse
was the fact that many "if" statements in this module didn't use
braces. Since the tracing macros add multiple statements, many "if"
statements had to be converted to use braces.
* Excessive use of PATH_MAX. Previous maintainers of this module
used PATH_MAX to allocate character arrays for filesystem paths
and SQL statements as though they cost nothing. In fact, PATH_MAX
is defined as 4096 bytes! Some functions had (and still have)
multiples of these. One function has 7. Given that the vast
majority of installations use the default spool directory path
`/var/spool/asterisk/voicemail`, the actual path length is usually
less than 80 bytes. That's over 4000 bytes wasted. It was the
same for SQL statement buffers. A 4K buffer for statement that
only needed 60 bytes. All of these PATH_MAX allocations in the
ODBC related code were changed to dynamically allocated buffers.
The rest will have to be addressed separately.
* Bug fixes. During the development of this feature, several
pre-existing ODBC related bugs were discovered and fixed. They
had to do with leaving orphaned files on disk, not preserving
original message ids when moving messages between folders,
not honoring the "formats" config parameter in certain circumstances,
etc.
UserNote: This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
(cherry picked from commit 2b9dc329bd)
From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition.
Resolves: https://github.com/asterisk/asterisk/issues/768
(cherry picked from commit a1d0dac6c6)
Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event
in the queue log if the Caller ID is restricted.
Resolves: #765
UpgradeNote: Add a new column to the queues table:
queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
to control whether the Restricted Caller ID will be stored in the queue log.
UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
will be stored in the queue log.
If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
(cherry picked from commit 624c23310e)
The current width for "extension" is 20 and "device state id" is 20, which is too small.
The "extension" field contains "ext"@"context", so 20 characters is not enough.
The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.
Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.
Resolves: #770
UserNote: The fields width of "core show hints" were increased.
The width of "extension" field to 30 characters and
the width of the "device state id" field to 60 characters.
(cherry picked from commit dcdda4688e)
Various SIP headers permit a URI to be prefaced with a `display-name`
production that can include characters (like commas and parentheses)
that are problematic for Asterisk's dialplan parser and, specifically
in the case of this patch, the PJSIP_PARSE_URI function.
This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
behaves identically to `PJSIP_PARSE_URI` except that the first
argument is now a variable name and not a literal URI.
Fixes#756
(cherry picked from commit 5068cc814f)
