asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	c5e30b7e15	Update for 20.9.3	9 months ago
George Joseph	a15050650a	res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback 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	9 months ago
Asterisk Development Team	81bfe0a62b	Update for 20.9.2	10 months ago
George Joseph	bbe68db10a	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	10 months ago
Asterisk Development Team	d0d75f61fe	Update for 20.9.1	11 months ago
George Joseph	95e5c90948	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	11 months ago
George Joseph	6b6025ac50	voicemail.conf.sample: Fix ':' comment typo ...and removed an errant trailing space. Resolves: #819	11 months ago
Asterisk Development Team	dc37086ccd	Update for 20.9.0	11 months ago
Asterisk Development Team	ae55f7f942	Update for 20.9.0-rc1	11 months ago
Igor Goncharovsky	19bbcf0bbf	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 `606df57b64`)	11 months ago
Mike Bradeen	e86bd1d97d	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 `d59f1a0c44`)	11 months ago
George Joseph	56b0c48afe	.github: Pass app_id and app_priv_key to AsteriskMergePR (cherry picked from commit `502d8a8811`)	11 months ago
George Joseph	524e7db2fd	.github: Change OnPRMergeApproved to use default token (cherry picked from commit `2350117539`)	11 months ago
George Joseph	fed053e83d	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 `59d75aea15`)	11 months ago
George Joseph	831219d74d	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 `a45e5ebf8d`)	11 months ago
Sean Bright	ff7dffcccd	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. Fixes #785 (cherry picked from commit `f3404d8377`)	11 months ago
George Joseph	2efe4011d5	.github: Use ASTERISKTEAM_PAT for PR merging (cherry picked from commit `17c3080945`)	11 months ago
George Joseph	e1706d33c1	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 `cbbf2891d2`)	11 months ago
George Joseph	a9f4bf7abb	.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 `b4eb773d5e`)	11 months ago
George Joseph	c972bb964a	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions (cherry picked from commit `d991b22dd8`)	11 months ago
George Joseph	0905dfdc1c	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 `e8c9cb8021`)	11 months ago
Tinet-mucw	a56925d9e6	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 `642df06a32`)	11 months ago
Alexei Gradinari	c3f78b36a0	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 `41d4db99cf`)	11 months ago
Alexei Gradinari	0831692f18	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 `75b550c8e1`)	11 months ago
Sean Bright	d3f48d67a9	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 `ceebc903ff`)	11 months ago
Sean Bright	411e9133b3	manager.c: Properly terminate `CoreShowChannelMap` event. Fixes #761 (cherry picked from commit `1ce5731c40`)	11 months ago
Bastian Triller	7e2a5fbd21	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 `8c152a7e7a`)	11 months ago
Sean Bright	5f8f4a6852	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 `21e3f84e56`)	11 months ago
chrsmj	b5b873d8b8	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 `c3d2f32852`)	11 months ago
Sean Bright	63e25ed4fe	asterisk.c: Don't log an error if .asterisk_history does not exist. Fixes #751 (cherry picked from commit `70f469139c`)	11 months ago
Walter Doekes	79673cf008	chan_ooh323: Fix R/0 typo in docs Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e (cherry picked from commit `4225475b50`)	11 months ago
Sean Bright	c618daeadd	bundled_pjproject: Disable UPnP support. Fixes #747 (cherry picked from commit `5088d35913`)	11 months ago
Sean Bright	73c702eca7	file.h: Rename function argument to avoid C++ keyword clash. Fixes #744 (cherry picked from commit `b769bee218`)	11 months ago
George Joseph	78a142db81	.github: Add branches to workflow_dispatch for NightlyTests (cherry picked from commit `c6e82783da`)	11 months ago
Mike Bradeen	728dbdccd1	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 `182ea91fc5`)	11 months ago
Asterisk Development Team	14367caaf7	Update for 20.8.1	1 year ago
George Joseph	741183d7ac	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	c98d80eaac	Update for 20.8.0	1 year ago
Asterisk Development Team	3e1b1560a9	Update for 20.8.0-rc1	1 year ago
Ivan Poddubny	c89b151fa8	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 `6079ea6d31`)	1 year ago
Ivan Poddubny	2893bfb2d6	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 `6c7e8876a5`)	1 year ago
Fabrice Fontaine	59aac533c8	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 `d452db3853`)	1 year ago
Naveen Albert	d46261b5e4	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 `001176c239`)	1 year ago
Naveen Albert	53aad00e6b	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 `261f3a373d`)	1 year ago
Naveen Albert	e5f9ce52ed	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 `c38f352d20`)	1 year ago
Naveen Albert	42ca53c6a9	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 `e634da7108`)	1 year ago
Henrik Liljedahl	e27f88ed03	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 `9cf8678112`)	1 year ago
George Joseph	c9439c8ad7	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 `9e1a6fa0a7`)	1 year ago
George Joseph	824b4ce855	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 `9c1b98e577`)	1 year ago
George Joseph	3bbdad0191	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 `af3dd9e2e3`)	1 year ago

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