asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	30c7925a0a	Update for 21.4.2	1 year ago
George Joseph	b4063bf756	manager.c: Add entries to Originate blacklist 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	1 year ago
Asterisk Development Team	a85124d963	Update for 21.4.1	1 year ago
George Joseph	1090fc1315	rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset() 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	1 year ago
George Joseph	5513a07969	voicemail.conf.sample: Fix ':' comment typo ...and removed an errant trailing space. Resolves: #819	1 year ago
Asterisk Development Team	e5362959e8	Update for 21.4.0	1 year ago
Asterisk Development Team	1b597be62a	Update for 21.4.0-rc1	1 year ago
Igor Goncharovsky	25d949b03f	res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS() 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 `aeefedb086`)	1 year ago
Mike Bradeen	56f1c20952	res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits 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 `7d53986262`)	1 year ago
George Joseph	31ff20988c	.github: Pass app_id and app_priv_key to AsteriskMergePR (cherry picked from commit `e0f7a5f968`)	1 year ago
George Joseph	af9c9f9ebe	.github: Change OnPRMergeApproved to use default token (cherry picked from commit `3b55311eb1`)	1 year ago
George Joseph	1ba2317472	ast-db-manage: Remove duplicate enum creation 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 `af58084855`)	1 year ago
George Joseph	4d5b30ca1c	security_agreement.c: Always add the Require and Proxy-Require headers 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 `210fe614b2`)	1 year ago
Sean Bright	21aa3d557f	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. Fixes #785 (cherry picked from commit `f55df3621a`)	1 year ago
George Joseph	8c874c6c20	.github: Use ASTERISKTEAM_PAT for PR merging (cherry picked from commit `a9b74ffa52`)	1 year ago
George Joseph	89d4f1015e	stasis_channels: Use uniqueid and name to delete old snapshots 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 `27f7cb6ea0`)	1 year ago
George Joseph	73c36d592e	.github: Replace PR workflows with stubs that call reusables The PR workflows now are just stubs that call reusable workflows located in the asterisk-ci-actions repo. (cherry picked from commit `b133383389`)	1 year ago
George Joseph	0e0be7070b	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions (cherry picked from commit `b867fa2fe5`)	1 year ago
George Joseph	e16d498ce7	app_voicemail_odbc: Allow audio to be kept on disk 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 `1b3a73cb24`)	1 year ago
Tinet-mucw	12193c70de	bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while iterating over bridge->channels. 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 `68a9c5683a`)	1 year ago
Alexei Gradinari	8a09a6ca6d	app_queue: Add option to not log Restricted Caller ID to queue_log 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 `192a848311`)	1 year ago
Alexei Gradinari	835d12ead2	pbx.c: expand fields width of "core show hints" 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 `8d7ee89047`)	1 year ago
Sean Bright	f88812ff86	pjsip: Add PJSIP_PARSE_URI_FROM dialplan function. 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 `78d63bc11c`)	1 year ago
Sean Bright	28c9d87837	manager.c: Properly terminate `CoreShowChannelMap` event. Fixes #761 (cherry picked from commit `7920e75da5`)	1 year ago
Bastian Triller	f01148396f	cli: Show configured cache dir Since Asterisk 19 it is possible to cache recorded files into another directory [1] [2]. Show configured location of cache dir in CLI's core show settings. [1] ASTERISK-29143 [2] `b08427134f` (cherry picked from commit `b56d50ba16`)	1 year ago
Sean Bright	44ed1cd50d	xml.c: Update deprecated libxml2 API usage. 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 (cherry picked from commit `f9a359c5c5`)	1 year ago
chrsmj	dd5ced4102	cdr_pgsql: Fix crash when the module fails to load multiple times. Missing or corrupt cdr_pgsql.conf configuration file can cause the second attempt to load the PostgreSQL CDR module to crash Asterisk via the Command Line Interface because a null CLI command is registered on the first failed attempt to load the module. Resolves: #736 (cherry picked from commit `735330bbd1`)	1 year ago
Sean Bright	791363d2e6	asterisk.c: Don't log an error if .asterisk_history does not exist. Fixes #751 (cherry picked from commit `fa043c8239`)	1 year ago
Walter Doekes	8aaa2802de	chan_ooh323: Fix R/0 typo in docs Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e (cherry picked from commit `f5a6ff5de5`)	1 year ago
Sean Bright	802a9902ed	bundled_pjproject: Disable UPnP support. Fixes #747 (cherry picked from commit `542cd37830`)	1 year ago
Sean Bright	581632538b	file.h: Rename function argument to avoid C++ keyword clash. Fixes #744 (cherry picked from commit `d51a86dc91`)	1 year ago
George Joseph	e5112eae8d	.github: Add branches to workflow_dispatch for NightlyTests (cherry picked from commit `12b997312b`)	1 year ago
Mike Bradeen	cf564713a4	rtp_engine: add support for multirate RFC2833 digits 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 (cherry picked from commit `624f509ce4`)	1 year ago
Asterisk Development Team	3b1125633e	Update for 21.3.1	1 year ago
George Joseph	22c3ff2b0c	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
Asterisk Development Team	726eac2f60	Update for 21.3.0	1 year ago
Asterisk Development Team	e96421ca94	Update for 21.3.0-rc1	1 year ago
Ivan Poddubny	cef2420083	configs: Fix a misleading IPv6 ACL example in Named ACLs "deny=::" is equivalent to "::/128". In order to mean "deny everything by default" it must be "::/0". (cherry picked from commit `685f525b28`)	1 year ago
Ivan Poddubny	9cc66fc175	asterisk.c: Fix sending incorrect messages to systemd notify Send "RELOADING=1" instead of "RELOAD=1" to follow the format expected by systemd (see sd_notify(3) man page). Do not send STOPPING=1 in remote console mode: attempting to execute "asterisk -rx" by the main process leads to a warning if NotifyAccess=main (the default) or to a forced termination if NotifyAccess=all. (cherry picked from commit `d3ff7c3eee`)	1 year ago
Fabrice Fontaine	942972b21c	res/stasis/control.c: include signal.h 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 (cherry picked from commit `c47307567a`)	1 year ago
Naveen Albert	f13130bd14	res_pjsip_logger: Preserve logging state on reloads. 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. (cherry picked from commit `9fc596aaa7`)	1 year ago
Naveen Albert	1214c14728	logger: Add unique verbose prefixes for levels 5-10. Add unique verbose prefixes for levels higher than 4, so that these can be visually differentiated from each other. Resolves: #721 (cherry picked from commit `57bb09667d`)	1 year ago
Naveen Albert	2b78e9a5fa	say.c: Fix cents off-by-one due to floating point rounding. Some of the money announcements can be off by one cent, due to the use of floating point in the money calculations, which is bad for obvious reasons. This replaces floating point with simple string parsing to ensure the cents value is converted accurately. Resolves: #525 (cherry picked from commit `3a49f9ade5`)	1 year ago
Naveen Albert	78ad353cc1	loader.c: Allow dependent modules to be unloaded recursively. Because of the (often recursive) nature of module dependencies in Asterisk, hot swapping a module on the fly is cumbersome if a module is depended on by other modules. Currently, dependencies must be popped manually by unloading dependents, unloading the module of interest, and then loading modules again in reverse order. To make this easier, the ability to do this recursively in certain circumstances has been added, as an optional extension to the "module refresh" command. If requested, Asterisk will check if a module that has a positive usecount could be unloaded safely if anything recursively dependent on it were unloaded. If so, it will go ahead and unload all these modules and load them back again. This makes hot swapping modules that provide dependencies much easier. Resolves: #474 UserNote: In certain circumstances, modules with dependency relations can have their dependents automatically recursively unloaded and loaded again using the "module refresh" CLI command or the ModuleLoad AMI command. (cherry picked from commit `a056e94885`)	1 year ago
Henrik Liljedahl	d1e5e4c4c1	res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting. 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 (cherry picked from commit `1423cfee29`)	1 year ago
George Joseph	c9d07a3f74	tcptls/iostream: Add support for setting SNI on client TLS connections If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello. (cherry picked from commit `4d6f84a14f`)	1 year ago
George Joseph	18c0cafa10	stir_shaken: Fix memory leak, typo in config, tn canonicalization * 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 (cherry picked from commit `b7ed77a7c5`)	1 year ago
George Joseph	8077e38a85	make_buildopts_h: Always include DETECT_DEADLOCKS Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must always be included in buildopts.h instead of only when ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise. Resolves: #719 (cherry picked from commit `d1d80af0c0`)	1 year ago
Spiridonov Dmitry	61ede2726f	sorcery.c: Fixed crash error when executing "module reload" Fixed crash error when cli "module reload". The error appears when compiling with res_prometheus and using the sorcery memory cache for registrations (cherry picked from commit `b8525e87ae`)	1 year ago
Naveen Albert	bd917229b4	callerid.c: Parse previously ignored Caller ID parameters. Commit `f2f397c1a8` previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681 (cherry picked from commit `4cf8d9d94a`)	1 year ago

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