33927 Commits (6b7b5c98dfa933c63f0550d33ee4dfcc9ffaa351)

 

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Author	SHA1	Message	Date
Sean Bright	6b7b5c98df	res_pjsip_logger.c: Fix 'OPTIONS' tab completion. ... Fixes #843 (cherry picked from commit b0e1f6a6ce)	1 year ago
Sean Bright	5a1e6995d6	alembic: Make 'revises' header comment match reality. ... (cherry picked from commit 53f2c426cc)	1 year ago
Mike Bradeen	31b8dc0923	Update version for Asterisk 22	1 year ago
Cade Parker	7f2ac6cb4f	chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay ... 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)	1 year ago
Mike Bradeen	b2455b2732	res_pjsip_notify: add dialplan application ... 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>	1 year ago
George Joseph	2328a0ab3a	manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE ... 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.	1 year ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... 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.	1 year ago
Jaco Kroon	3f1d1a1463	configure: Use . file rather than source file. ... 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>	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event ... 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.	1 year ago
George Joseph	0f67c2696f	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
Mike Bradeen	a824a0cdb4	res_stasis: fix intermittent delays on adding channel to bridge ... 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	1 year ago
Tinet-mucw	291c1e4c1f	res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto ... 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	1 year ago
George Joseph	b7879224c1	.github: Allow testing an Asterisk PR against a testsuite PR	1 year ago
George Joseph	e4b0ad45e9	.github: Add params to Releaser for FPBX issue creation	1 year ago
George Joseph	cee50bab7a	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	a02fc685a8	stir_shaken: CRL fixes and a new CLI command ... * 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	1 year ago
George Joseph	f45f8781d0	res_pjsip_config_wizard.c: Refactor load process ... 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.	1 year ago
George Joseph	79e0b50162	voicemail.conf.sample: Fix ':' comment typo ... ...and removed an errant trailing space. Resolves: #819	1 year ago
George Joseph	bc05416004	bridge_softmix: Fix queueing VIDUPDATE control frames ... 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	1 year ago
Igor Goncharovsky	55f4e6dd03	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.	1 year ago
Mike Bradeen	52bdfe09ab	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	1 year ago
George Joseph	c0d8831636	.github: Pass app_id and app_priv_key to AsteriskMergePR	1 year ago
George Joseph	6cce75409e	.github: Change OnPRMergeApproved to use default token	1 year ago
George Joseph	e6633e8fdd	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	1 year ago
George Joseph	0092dd62c5	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	1 year ago
Sean Bright	8b5ddfee5e	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. ... Fixes #785	1 year ago
George Joseph	b8c440a52f	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	1 year ago
George Joseph	e5b6a5be5e	.github: Use ASTERISKTEAM_PAT for PR merging	1 year ago
George Joseph	6feaed5f94	.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.	1 year ago
George Joseph	9076d3678d	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions	1 year ago
George Joseph	6e73af341a	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.	1 year ago
Tinet-mucw	7d28165cb1	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	1 year ago
Alexei Gradinari	07aa0ffe6a	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.	1 year ago
Alexei Gradinari	4280d7cd77	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.	1 year ago
Sean Bright	25686a57da	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	1 year ago
Sean Bright	26eabed246	manager.c: Properly terminate `CoreShowChannelMap` event. ... Fixes #761	1 year ago
Bastian Triller	843cc8f2cd	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	1 year ago
Sean Bright	f3e88d366c	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	1 year ago
chrsmj	f789b7765b	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	1 year ago
Sean Bright	e1db3027af	asterisk.c: Don't log an error if .asterisk_history does not exist. ... Fixes #751	1 year ago
Walter Doekes	d065895c83	chan_ooh323: Fix R/0 typo in docs ... Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e	1 year ago
Sean Bright	715d1b5d43	bundled_pjproject: Disable UPnP support. ... Fixes #747	1 year ago
Sean Bright	f3449f3370	file.h: Rename function argument to avoid C++ keyword clash. ... Fixes #744	1 year ago
George Joseph	85241bd229	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
George Joseph	115d7c01e3	.github: Add branches to workflow_dispatch for NightlyTests	1 year ago
Mike Bradeen	3b4d2a6506	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	1 year ago
Ivan Poddubny	abe0018d85	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".	1 year ago
Ivan Poddubny	29675e3aab	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.	1 year ago
Fabrice Fontaine	7fc25285fa	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	1 year ago
Naveen Albert	077962dd7b	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.	1 year ago