asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	faddd99f2b	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	6ce8fdc32c	Update for 18.24.1	1 year ago
George Joseph	275e17a663	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	39128ddf8c	voicemail.conf.sample: Fix ':' comment typo ...and removed an errant trailing space. Resolves: #819	1 year ago
Asterisk Development Team	78c256c939	Update for 18.24.0	1 year ago
Asterisk Development Team	615ea039c4	Update for 18.24.0-rc1	1 year ago
Igor Goncharovsky	38aab7a9ad	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 `caa1820d2b`)	1 year ago
Mike Bradeen	9fc2ada28c	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 `ac9c510d99`)	1 year ago
George Joseph	1b702a396b	.github: Pass app_id and app_priv_key to AsteriskMergePR (cherry picked from commit `ff31908906`)	1 year ago
George Joseph	96a8062d85	.github: Change OnPRMergeApproved to use default token (cherry picked from commit `947dc2cd78`)	1 year ago
George Joseph	f820ee4b59	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 `614d71fa19`)	1 year ago
George Joseph	8fc3df467e	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 `1d9f43b5a5`)	1 year ago
Sean Bright	9f9a6cf46e	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. Fixes #785 (cherry picked from commit `0bcee9de5d`)	1 year ago
George Joseph	54810a8370	.github: Use ASTERISKTEAM_PAT for PR merging (cherry picked from commit `df8fbe222a`)	1 year ago
George Joseph	336ba360bd	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 `6f67835b4f`)	1 year ago
George Joseph	be6d17202c	.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 `2cb20b4bac`)	1 year ago
George Joseph	c2c24114ba	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions (cherry picked from commit `08080d3b60`)	1 year ago
George Joseph	c9d5fd450e	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 `2b9dc329bd`)	1 year ago
Tinet-mucw	c759489d3e	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 `a1d0dac6c6`)	1 year ago
Alexei Gradinari	1108f61bfb	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 `624c23310e`)	1 year ago
Alexei Gradinari	e93c5e945a	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 `dcdda4688e`)	1 year ago
Sean Bright	d18159a008	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 `5068cc814f`)	1 year ago
Sean Bright	9afc995824	manager.c: Properly terminate `CoreShowChannelMap` event. Fixes #761 (cherry picked from commit `87278c087a`)	1 year ago
Sean Bright	a2980fe5fa	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 `126cb5a20d`)	1 year ago
chrsmj	9f369fb5fd	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 `4fea8a433b`)	1 year ago
Sean Bright	877d5ff577	asterisk.c: Don't log an error if .asterisk_history does not exist. Fixes #751 (cherry picked from commit `e9fb27f804`)	1 year ago
Walter Doekes	823c948842	chan_ooh323: Fix R/0 typo in docs Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e (cherry picked from commit `4f998eb5aa`)	1 year ago
Sean Bright	6042ce4a78	bundled_pjproject: Disable UPnP support. Fixes #747 (cherry picked from commit `ff510c644a`)	1 year ago
Sean Bright	f0b48a63b6	file.h: Rename function argument to avoid C++ keyword clash. Fixes #744 (cherry picked from commit `9942ebae86`)	1 year ago
George Joseph	056d83043e	.github: Add branches to workflow_dispatch for NightlyTests (cherry picked from commit `d2456e0436`)	1 year ago
Mike Bradeen	992681cc45	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 `6bf66b82d7`)	1 year ago
Asterisk Development Team	351c871038	Update for 18.23.1	1 year ago
George Joseph	3572cc8f4b	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	14c4343b72	Update for 18.23.0	1 year ago
Asterisk Development Team	ccebe76dcd	Update for 18.23.0-rc1	1 year ago
Ivan Poddubny	5ae0c186aa	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 `ee6a0365ab`)	1 year ago
Ivan Poddubny	3988aff68e	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 `3e9922d389`)	1 year ago
Fabrice Fontaine	9efb9c7d45	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 `044f3e85ba`)	1 year ago
Naveen Albert	fc35e92ece	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 `3b600e2787`)	1 year ago
Naveen Albert	1acf473c5b	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 `23f5ce69fd`)	1 year ago
Naveen Albert	62f9c54f4c	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 `b836f9c01c`)	1 year ago
Naveen Albert	46ddecb3d5	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 `c276ae11e0`)	1 year ago
Henrik Liljedahl	f580b8b5f1	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 `d74266fe05`)	1 year ago
George Joseph	229f3035f0	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 `f9a1e3095c`)	1 year ago
George Joseph	7d1a8e8411	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 `3d2def92e2`)	1 year ago
George Joseph	45eec15df1	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 `6568c86989`)	1 year ago
Spiridonov Dmitry	f817526226	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 `5a883f5410`)	1 year ago
Naveen Albert	a2579ec402	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 `6bd0b67081`)	1 year ago
George Joseph	6e8678d8ac	logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT If you're tracing a large function that may call another function multiple times in different circumstances, it can be difficult to see from the trace output exactly which location that function was called from. There's no good way to automatically determine the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT simply print out a trace line before and after the call. The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is that SCOPE_CALL ignores the function's return value (if any) where SCOPE_CALL_WITH_RESULT allows you to specify the type of the function's return value so it can be assigned to a variable. SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT and the "int" return type. (cherry picked from commit `65018e8ebf`)	1 year ago
Sean Bright	e49049b474	app_queue.c: Properly handle invalid strategies from realtime. The existing code sets the queue strategy to `ringall` but it is then immediately overwritten with an invalid one. Fixes #707 (cherry picked from commit `574d6986ce`)	1 year ago

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