33455 Commits (49cfdbbdff315e5a6bb665bb44a09d881f71b52c)

 

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Author	SHA1	Message	Date
Naveen Albert	49cfdbbdff	manager: Update ModuleCheck documentation. ... The ModuleCheck XML documentation falsely claims that the module's version number is returned. This has not been the case since 14, since the version number is not available anymore, but the documentation was not changed at the time. It is now updated to reflect this. ASTERISK-30285 #close Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91	4 years ago
Naveen Albert	8142b313c3	file.c: Don't emit warnings on winks. ... Adds an ignore case for wink since it should pass through with no warning. ASTERISK-30290 #close Change-Id: Ieb7e34daa717357ac5c93efb0059f6c2321f16ad	4 years ago
George Joseph	0c1c623dee	runUnittests.sh: Save coredumps to proper directory ... Fixed the specification of "outputdir" when calling ast_coredumper so the txt files are saved in the correct place. ASTERISK-30282 Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6	4 years ago
Naveen Albert	dfe2f38642	app_stack: Print proper exit location for PBXless channels. ... When gosub is executed on channels without a PBX, the context, extension, and priority are initialized to the channel driver's default location for that endpoint. As a result, the last Return will restore this location and the Gosub logs will print out bogus information about our exit point. To fix this, on channels that don't have a PBX, the execution location is left intact on the last return if there are no further stack frames left. This allows the correct location to be printed out to the user, rather than the bogus default context. ASTERISK-30076 #close Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414	4 years ago
George Joseph	f723b465e5	chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer ... unicast_rtp_request() was setting the channel variables like this: pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address)); ast_rtp_instance_get_local_address(instance, &local_address); pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address)); ...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being set before local_address was set. In fact, the address part of local_address was set earlier in the function, just not the port. This was confusing however so ast_rtp_instance_get_local_address() is now being called before setting UNICASTRTP_LOCAL_ADDRESS. ASTERISK-30281 Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4	4 years ago
Mike Bradeen	50e2921a48	res_pjsip: prevent crash on websocket disconnect ... When a websocket (or potentially any stateful connection) is quickly created then destroyed, it is possible that the qualify thread will destroy the transaction before the initialzing thread is finished with it. Depending on the timing, this can cause an assertion within pjsip. To prevent this, ast_send_stateful_response will now create the group lock and add a reference to it before creating the transaction. While this should resolve the crash, there is still the potential that the contact will not be cleaned up properly, see:ASTERISK~29286. As a result, the contact has to 'time out' before it will be removed. ASTERISK-28689 Change-Id: Id050fded2247a04d8f0fc5b8a2cf3e5482cb8cee	4 years ago
Naveen Albert	afd86b47c1	tcptls: Prevent crash when freeing OpenSSL errors. ... write_openssl_error_to_log has been erroneously using ast_free instead of free, which will cause a crash when MALLOC_DEBUG is enabled since the memory was not allocated by Asterisk's memory manager. This changes it to use the actual free function directly to avoid this. ASTERISK-30278 #close Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d	4 years ago
Igor Goncharovsky	096529d33f	res_pjsip_outbound_registration: Allow to use multiple proxies for registration ... Current registration code use pjsip_parse_uri to verify outbound_proxy that is different from the reading this option for the endpoint. This made value with multiple proxies invalid for registration pjsip settings. Removing URI validation helps to use registration through multiple proxies. ASTERISK-30217 #close Change-Id: I064558e66f04b9f3260c46181812a01349761357	4 years ago
Naveen Albert	ca8900b0f6	tests: Fix compilation errors on 32-bit. ... Fix compilation errors caused by using size_t instead of uintmax_t and non-portable format specifiers. ASTERISK-30273 #close Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee	4 years ago
Henning Westerholt	12445040d3	res_pjsip: return all codecs on a re-INVITE without SDP ... Currently chan_pjsip on receiving a re-INVITE without SDP will only return the codecs that are previously negotiated and not offering all enabled codecs. This causes interoperability issues with different equipment (e.g. from Cisco) for some of our customers and probably also in other scenarios involving 3PCC infrastructure. According to RFC 3261, section 14.2 we SHOULD return all codecs on a re-INVITE without SDP The PR proposes a new parameter to configure this behaviour: all_codecs_on_empty_reinvite. It includes the code, documentation, alembic migrations, CHANGES file and example configuration additions. ASTERISK-30193 #close Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148	4 years ago
Naveen Albert	40b52322e5	res_pjsip_notify: Add option support for AMI. ... The PJSIP notify CLI commands allow for using "options" configured in pjsip_notify.conf. This allows these same options to be used in AMI actions as well. Additionally, as part of this improvement, some repetitive common code is refactored. ASTERISK-30263 #close Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5	4 years ago
Naveen Albert	c32b39d123	res_pjsip_logger: Add method-based logging option. ... Expands the pjsip logger to support the ability to filter by SIP message method. This can make certain types of SIP debugging easier by only logging messages of particular method(s). ASTERISK-30146 #close Co-authored-by: Sean Bright <sean@seanbright.com> Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f	4 years ago
Frederic LE FOLL	50a4495799	Dialing API: Cancel a running async thread, may not cancel all calls ... race condition: ast_dial_join() may not cancel outgoing call, if function is called just after called party answer and before application execution (bit is_running_app not yet set). This fix adds ast_softhangup() calls in addition to existing pthread_kill() when is_running_app is not set. ASTERISK-30258 Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4	4 years ago
Naveen Albert	180ca32565	chan_dahdi: Fix unavailable channels returning busy. ... This fixes dahdi_request to properly set the cause code to CONGESTION instead of BUSY if no channels were actually available. Currently, the cause is erroneously set to busy if the channel itself is found, regardless of its current state. However, if the channel is not available (e.g. T1 down, card not operable, etc.), then the channel itself may not be in a functional state, in which case CHANUNAVAIL is the correct cause to use. This adds a simple check to ensure that busy tone is only returned if a channel is encountered that has an owner, since that is the only possible way that a channel could actually be busy. ASTERISK-30274 #close Change-Id: Iad5870223c081240c925b19df8d6af136953b994	4 years ago
Naveen Albert	9258d8212a	res_pjsip_pubsub: Prevent removing subscriptions. ... pjproject does not provide any mechanism of removing event packages, which means that once a subscription handler is registered, it is effectively permanent. pjproject will assert if the same event package is ever registered again, so currently unloading and loading any Asterisk modules that use subscriptions will cause a crash that is beyond our control. For that reason, we now prevent users from being able to unload these modules, to prevent them from ever being loaded twice. ASTERISK-30264 #close Change-Id: I7fdcb1a5e44d38b7ba10c44259fe98f0ae9bc12c	4 years ago
Naveen Albert	407216a0a5	say: Don't prepend ampersand erroneously. ... Some logic in say.c for determining if we need to also add an ampersand for file seperation was faulty, as non-successful files would increment the count, causing a leading ampersand to be added improperly. This is fixed, and a unit test that captures this regression is also added. ASTERISK-30248 #close Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b	4 years ago
Philip Prindeville	d0bea5a725	res_crypto: handle unsafe private key files ... ASTERISK-30213 #close Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a	4 years ago
Mike Bradeen	907d7e7d7d	audiohook: add directional awareness ... Add enum to allow setting optional direction. If set to only one direction, only feed matching-direction frames to the associated slin factory. This prevents mangling the transcoder on non-mixed frames when the READ and WRITE frames would have otherwise required it. Also removes the need to mute or discard the un-wanted frames as they are no longer added in the first place. res_stasis_snoop is changed to use this addition to set direction on audiohook based on spy direction. If no direction is set, the ast_audiohook_init will init this enum to BOTH which maintains existing functionality. ASTERISK-30252 Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb	4 years ago
Naveen Albert	b331caca30	cdr: Allow bridging and dial state changes to be ignored. ... Allows bridging, parking, and dial messages to be globally ignored for all CDRs such that only a single CDR record is generated per channel. This is useful when CDRs should endure for the lifetime of an entire channel and bridging and dial updates in the dialplan should not result in multiple CDR records being created for the call. With the ignore bridging option, bridging changes have no impact on the channel's CDRs. With the ignore dial state option, multiple Dials and their outcomes have no impact on the channel's CDRs. The last disposition on the channel is preserved in the CDR, so the actual disposition of the call remains available. These two options can reduce the amount of "CDR hacks" that have hitherto been necessary to ensure that CDR was not "spoiled" by these messages if that was undesired, such as putting a dummy optimization-disabled local channel between the caller and the actual call and putting the CDR on the channel in the middle to ensure that CDR would persist for the entire call and properly record start, answer, and end times. Enabling these options is desirable when calls correspond to the entire lifetime of channels and the CDR should reflect that. Current default behavior remains unchanged. ASTERISK-30091 #close Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832	4 years ago
Naveen Albert	e0e7f35730	res_tonedetect: Add ringback support to TONE_DETECT. ... Adds support for detecting audible ringback tone to the TONE_DETECT function using the p option. ASTERISK-30254 #close Change-Id: Ie2329ff245248768367d26749c285fbe823f6414	4 years ago
Naveen Albert	98fc05f13b	chan_dahdi: Resolve format truncation warning. ... Fixes a format truncation warning in notify_message. ASTERISK-30256 #close Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1	4 years ago
Philip Prindeville	ef74ecacc7	res_crypto: don't modify fname in try_load_key() ... "fname" is passed in as a const char *, but strstr() mangles that into a char *, and we were attempting to modify the string in place. This is an unwanted (and undocumented) side-effect. ASTERISK-30213 Change-Id: Ifa36d352aafeb7f9beec3f746332865c7d21e629	4 years ago
Philip Prindeville	5e2485b5c0	res_crypto: use ast_file_read_dirs() to iterate ... ASTERISK-30213 Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0	4 years ago
George Joseph	2a500b325a	res_geolocation: Update wiki documentation ... Also added a note to the geolocation.conf.sample file and added a README to the res/res_geolocation/wiki directory. Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc	4 years ago
Maximilian Fridrich	0d2e140123	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	4 years ago
Asterisk Development Team	7f80830ced	Update CHANGES and UPGRADE.txt for 20.0.0	4 years ago
Holger Hans Peter Freyther	62881c668b	res_prometheus: Do not crash on invisible bridges ... Avoid crashing by skipping invisible bridges and checking the snapshot for a null pointer. In effect this is how the bridges are enumerated in res/ari/resource_bridges.c already. ASTERISK-30239 ASTERISK-30237 Change-Id: I58ef9f44036feded5966b5fc70ae754f8182883d	4 years ago
Naveen Albert	8afb313a43	res_pjsip_geolocation: Change some notices to debugs. ... If geolocation is not in use for an endpoint, the NOTICE log level is currently spammed with messages about this, even though nothing is wrong and these messages provide no real value. These log messages are therefore changed to debugs. ASTERISK-30241 #close Change-Id: I656b355d812f67cc0f0fdf09b00b0e1458598bb4	4 years ago
Naveen Albert	7335b0cffe	db: Fix incorrect DB tree count for AMI. ... The DBGetTree AMI action's ListItem previously always reported 1, regardless of the count. This is corrected to report the actual count. ASTERISK-30245 #close patches: gettreecount.diff submitted by Birger Harzenetter (license 5870) Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4	4 years ago
Naveen Albert	407167cc28	func_logic: Don't emit warning if both IF branches are empty. ... The IF function currently emits warnings if both IF branches are empty. However, there is no actual necessity that either branch be non-empty as, unlike other conditional applications/ functions, nothing is inherently done with IF, and both sides could legitimately be empty. The warning is thus turned into a debug message. ASTERISK-30243 #close Change-Id: I5250625dd720f95e1859b5dfb933905d7e7a730e	4 years ago
Naveen Albert	a5ec60e6c6	features: Add no answer option to Bridge. ... Adds the n "no answer" option to the Bridge application so that answer supervision can not automatically be provided when Bridge is executed. Additionally, a mechanism (dialplan variable) is added to prevent bridge targets (typically the target of a masquerade) from answering the channel when they enter the bridge. ASTERISK-30223 #close Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d	4 years ago
Naveen Albert	1e29607b5c	app_bridgewait: Add option to not answer channel. ... Adds the n option to not answer the channel when calling BridgeWait, so the application can be used without forcing answer supervision. ASTERISK-30216 #close Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a	4 years ago
Naveen Albert	8c791f9a65	app_amd: Add option to play audio during AMD. ... Adds an option that will play an audio file to the party while AMD is running on the channel, so the called party does not just hear silence. ASTERISK-30179 #close Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6	4 years ago
Philip Prindeville	3e7ce90f9c	test: initialize capture structure before freeing ... ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	4 years ago
Naveen Albert	1ed4518328	func_export: Add EXPORT function ... Adds the EXPORT function, which allows write access to variables and functions on other channels. ASTERISK-29432 #close Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf	4 years ago
Maximilian Fridrich	5bbad0d27c	res_pjsip: Add 100rel option "peer_supported". ... This patch adds a new option to the 100rel parameter for pjsip endpoints called "peer_supported". When an endpoint with this option receives an incoming request and the request indicated support for the 100rel extension, then Asterisk will send 1xx responses reliably. If the request did not indicate 100rel support, Asterisk sends 1xx responses normally. ASTERISK-30158 Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad	4 years ago
Naveen Albert	8aae0b9f08	func_scramble: Fix null pointer dereference. ... Fix segfault due to null pointer dereference inside the audiohook callback. ASTERISK-30220 #close Change-Id: Ideb80f606974366e89d619d908744230b5a5a259	4 years ago
Jaco Kroon	278c5726ca	manager: be more aggressive about purging http sessions. ... If we find that n_max (currently hard wired to 1) sessions were purged, schedule the next purge for 1ms into the future rather than 5000ms (as per current). This way we will purge up to 1000 sessions per second rather than 1 every 5 seconds. This mitigates a build-up of sessions should http sessions gets established faster than 1 per 5 seconds. Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Naveen Albert	ab1dbfef75	func_strings: Add trim functions. ... Adds TRIM, LTRIM, and RTRIM, which can be used for trimming leading and trailing whitespace from strings. ASTERISK-30222 #close Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554	4 years ago
George Joseph	e25b690d10	res_crypto: Memory issues and uninitialized variable errors ... ASTERISK-30235 Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd	4 years ago
George Joseph	e33f2dcc0f	res_geolocation: Fix issues exposed by compiling with -O2 ... Fixed "may be used uninitialized" errors in geoloc_config.c. ASTERISK-30234 Change-Id: I1ea336bf7abbc16fa59b75720f0db8f1d960b3d4	4 years ago
Philip Prindeville	026dc08529	res_crypto: don't complain about directories ... ASTERISK-30226 #close Change-Id: I5695fb0c9521f112f754b8362cff2a8f3eff05c5	4 years ago
Asterisk Development Team	f01ed3eea4	Update CHANGES and UPGRADE.txt for 20.0.0	4 years ago
Mike Bradeen	7a44296ca9	res_pjsip: Add user=phone on From and PAID for usereqphone=yes ... Adding user=phone to local-side uri's when user_eq_phone=yes is set for an endpoint. Previously this would only add the header to the To and R-URI. ASTERISK-30178 Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600	4 years ago
George Joseph	8cbea1c7ef	res_geolocation: Fix segfault when there's an empty element ... Fixed a segfault caused by var_list_from_loc_info() encountering an empty location info element. Fixed an issue in ast_strsep() where a value with only whitespace wasn't being preserved. Fixed an issue in ast_variable_list_from_quoted_string() where an empty value was considered a failure. ASTERISK-30215 Reported by: Dan Cropp Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a	4 years ago
sungtae kim	80bc844fd6	res_musiconhold: Add option to not play music on hold on unanswered channels ... This change adds an option, answeredonly, that will prevent music on hold on channels that are not answered. ASTERISK-30135 Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08	4 years ago
Ben Ford	881a3f2306	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	4 years ago
Philip Prindeville	3e054c9ebc	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	4 years ago
Philip Prindeville	736cdf84f4	test: Add coverage for res_crypto ... We're validating the following functionality: encrypting a block of data with RSA decrypting a block of data with RSA signing a block of data with RSA verifying a signature with RSA encrypting a block of data with AES-ECB encrypting a block of data with AES-ECB as well as accessing test keys from the keystore. ASTERISK-30045 #close Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d	4 years ago
Philip Prindeville	2d7656cb50	res_crypto: make keys reloadable on demand for testing ... ASTERISK-30045 Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8	4 years ago