asterisk

Commit Graph

Author	SHA1	Message	Date
Mike Bradeen	eb1d7ab53c	ooh323c: not checking for IE minimum length When decoding q.931 encoded calling/called number now checking for length being less than minimum required. ASTERISK-30103 Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7	2 years ago
Naveen Albert	c7df5ee7c1	pbx_builtins: Allow Answer to return immediately. The Answer application currently waits for up to 500ms for media, even if users specify a different timeout. This adds an option to not wait for media on the channel by doing a raw answer instead. The default 500ms threshold is also documented. ASTERISK-30308 #close Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4	2 years ago
Naveen Albert	5ede4e217a	chan_dahdi: Allow FXO channels to start immediately. Currently, chan_dahdi will wait for at least one ring before an incoming call can enter the dialplan. This is generally necessary in order to receive the Caller ID spill and/or distinctive ringing detection. However, if neither of these is required, then there is nothing gained by waiting for one ring and this unnecessarily delays call setup. Users can now use immediate=yes to make FXO channels (FXS signaled) begin processing dialplan as soon as Asterisk receives the call. ASTERISK-30305 #close Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5	2 years ago
Maximilian Fridrich	60b81eabe0	core & res_pjsip: Improve topology change handling. This PR contains two relatively separate changes in channel.c and res_pjsip_session.c which ensure that topology changes are not ignored in cases where they should be handled. For channel.c: The function ast_channel_request_stream_topology_change only triggers a stream topology request change indication, if the channel's topology does not equal the requested topology. However, a channel could be in a state where it is currently "negotiating" a new topology but hasn't updated it yet, so the topology request change would be lost. Channels need to be able to handle such situations internally and stream topology requests should therefore always be passed on. In the case of chan_pjsip for example, it queues a session refresh (re-INVITE) if it is currently in the middle of a transaction or has pending requests (among other reasons). Now, ast_channel_request_stream_topology_change always indicates a stream topology request change even if the requested topology equals the channel's topology. For res_pjsip_session.c: The function resolve_refresh_media_states does not process stream state changes if the delayed active state differs from the current active state. I.e. if the currently active stream state has changed between the time the sip session refresh request was queued and the time it is being processed, the session refresh is ignored. However, res_pjsip_session contains logic that ensures that session refreshes are queued and re-queued correctly if a session refresh is currently not possible. So this check is not necessary and led to some session refreshes being lost. Now, a session refresh is done even if the delayed active state differs from the current active state and it is checked whether the delayed pending state differs from the current active - because that means a refresh is necessary. Further, the unit test of resolve_refresh_media_states was adapted to reflect the new behavior. I.e. the changes to delayed pending are prioritized over the changes to current active because we want to preserve the original intention of the pending state. ASTERISK-30184 Change-Id: Icd0703295271089057717006730b555b9a1d4e5a	2 years ago
Naveen Albert	2efa290d3c	sla: Prevent deadlock and crash due to autoservicing. SLAStation currently autoservices the station channel before creating a thread to actually dial the trunk. This leads to duplicate servicing of the channel which causes assertions, deadlocks, crashes, and moreover not the correct behavior. Removing the autoservice prevents the crash, but if the station hangs up before the trunk answers, the call hangs since the hangup was never serviced on the channel. This is fixed by not autoservicing the channel, but instead servicing it in the thread dialing the trunk, since it is doing so synchronously to begin with. Instead of sleeping for 100ms in a loop, we simply use the channel for timing, and abort if it disappears. The same issue also occurs with SLATrunk when a call is answered, because ast_answer invokes ast_waitfor_nandfds. Thus, we use ast_raw_answer instead which does not cause any conflict and allows the call to be answered normally without thread blocking issues. ASTERISK-29998 #close Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8	2 years ago
Jaco Kroon	ce2153fc5a	Build system: Avoid executable stack. Found in res_geolocation, but I believe others may have similar issues, thus not linking to a specific issue. Essentially gcc doesn't mark the stack for being non-executable unless it's compiling the source, this informs ld via gcc to mark the object as not requiring an executable stack (which a binary blob obviously doesn't). ASTERISK-30321 Change-Id: I71bcc2fd1fe0c82a28b3257405d6f2b566fd9bfc Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Naveen Albert	002afc3f2a	func_json: Fix memory leak. A memory leak was present in func_json due to using ast_json_free, which just calls ast_free, as opposed to recursively freeing the JSON object as needed. This is now fixed to use the right free functions. ASTERISK-30293 #close Change-Id: I982324dde841dc9147c8d8ad35c8719daf418b49	2 years ago
Naveen Albert	1e77b8c473	test_json: Remove duplicated static function. Removes the function mkstemp_file and uses ast_file_mkftemp from file.h instead. ASTERISK-30295 #close Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609	2 years ago
Joshua C. Colp	61922d2934	res_agi: Respect "transmit_silence" option for "RECORD FILE". The "RECORD FILE" command in res_agi has its own implementation for actually doing the recording. This has resulted in it not actually obeying the option "transmit_silence" when recording. This change causes it to now send silence if the option is enabled. ASTERISK-30314 Change-Id: Ib3a85601ff35d1b904f836691bad8a4b7e957174	2 years ago
Naveen Albert	6e59b01e1a	app_mixmonitor: Add option to delete files on exit. Adds an option that allows MixMonitor to delete its copy of any recording files before exiting. This can be handy in conjunction with options like m, which copy the file elsewhere, and the original files may no longer be needed. ASTERISK-30284 #close Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e	3 years ago
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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 years ago
Philip Prindeville	d0bea5a725	res_crypto: handle unsafe private key files ASTERISK-30213 #close Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a	3 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	3 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	3 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	3 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	3 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	3 years ago
Philip Prindeville	5e2485b5c0	res_crypto: use ast_file_read_dirs() to iterate ASTERISK-30213 Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0	3 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	3 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	3 years ago
Asterisk Development Team	7f80830ced	Update CHANGES and UPGRADE.txt for 20.0.0	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 years ago
Philip Prindeville	3e7ce90f9c	test: initialize capture structure before freeing ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	3 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	3 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	3 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	3 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>	3 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	3 years ago
George Joseph	e25b690d10	res_crypto: Memory issues and uninitialized variable errors ASTERISK-30235 Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd	3 years ago

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