asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	d33a05bc75	Update for 19.8.1	2 years ago
Sean Bright	86da1af5ae	apply_patches: Use globbing instead of file/sort. This accomplishes the same thing as a `find ... \| sort` but with the added benefit of clarity and avoiding a call to a subshell. Additionally drop the -s option from call to patch as it is not POSIX.	2 years ago
George Joseph	f43849b65b	bundled_pjproject: Backport 2 SSL patches from upstream * Fix double free of ossock->ossl_ctx in case of errors https://github.com/pjsip/pjproject/commit/863629bc65d6 * free SSL context and reset context pointer when setting the cipher list fails https://github.com/pjsip/pjproject/commit/0fb32cd4c0b2 Resolves: #194	2 years ago
George Joseph	6e5e2cd526	bundled_pjproject: Backport security fixes from pjproject 2.13.1 Merge-pull-request-from-GHSA-9pfh-r8x4-w26w.patch Merge-pull-request-from-GHSA-cxwq-5g9x-x7fr.patch Locking-fix-so-that-SSL_shutdown-and-SSL_write-are-n.patch Don-t-call-SSL_shutdown-when-receiving-SSL_ERROR_SYS.patch Resolves: #188	2 years ago
George Joseph	ebdf660a56	apply_patches: Sort patch list before applying The apply_patches script wasn't sorting the list of patches in the "patches" directory before applying them. This left the list in an indeterminate order. In most cases, the list is actually sorted but rarely, they can be out of order and cause dependent patches to fail to apply. We now sort the list but the "sort" program wasn't in the configure scripts so we needed to add that and regenerate the scripts as well. Resolves: #193	2 years ago
Asterisk Development Team	e528299031	Update for 19.8.0	2 years ago
Asterisk Development Team	b40f0f6fce	Update for 19.8.0-rc2	2 years ago
George Joseph	7129752201	res_pjsip_transport_websocket: Add remote port to transport When Asterisk receives a new websocket conenction, it creates a new pjsip transport for it and copies connection data into it. The transport manager then uses the remote IP address and port on the transport to create a monitor for each connection. However, the remote port wasn't being copied, only the IP address which meant that the transport manager was creating only 1 monitoring entry for all websocket connections from the same IP address. Therefore, if one of those connections failed, it deleted the transport taking all the the connections from that same IP address with it. * We now copy the remote port into the created transport and the transport manager behaves correctly. ASTERISK-30369 Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1	2 years ago
Asterisk Development Team	1eee96543b	Update for 19.8.0-rc1	2 years ago
Asterisk Development Team	c4703f070a	Update CHANGES and UPGRADE.txt for 19.8.0	2 years ago
Alexandre Fournier	c900a7dc39	res_geoloc: fix NULL pointer dereference bug The `ast_geoloc_datastore_add_eprofile` function does not return 0 on success, it returns the size of the underlying datastore. This means that the datastore will be freed and its pointer set to NULL when no error occured at all. ASTERISK-30346 Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9	2 years ago
Ben Ford	1c8f57e298	pjproject: 2.13 security fixes Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13. ASTERISK-30338 Change-Id: I86fdc003d5d22cb66e7cc6dc3313a8194f27eb69	2 years ago
George Joseph	69c2459c33	pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	2 years ago
Mike Bradeen	764ca33473	manager: prevent file access outside of config dir Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404	2 years ago
Mike Bradeen	42ecf402a3	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	e553546eff	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
Mike Bradeen	d7dae7b1fc	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
Igor Goncharovsky	09af7e0aca	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	025dbcfd68	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	a6aebab79b	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	05a7b4132a	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
Naveen Albert	a0dd8c27b5	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
Frederic LE FOLL	83498bec35	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	3bae133afa	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	8bea5052db	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	57b1f5a7c3	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	23a4135fe7	res_crypto: handle unsafe private key files ASTERISK-30213 #close Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a	3 years ago
Mike Bradeen	f61dbd566b	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	b397dc3ca3	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	f9297117d9	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	11d97dc5ca	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	44684fdd51	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	109c2335ef	res_crypto: use ast_file_read_dirs() to iterate ASTERISK-30213 Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0	3 years ago
George Joseph	54cafbc67f	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	6170073800	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	2209afddb9	Update CHANGES and UPGRADE.txt for 19.7.0	3 years ago
George Joseph	95a25fddac	manager.h: Bump version to 8.0.2 Change-Id: I99c16241f74a909c792cec3c722a3142a1892197	3 years ago
Holger Hans Peter Freyther	02be2a5f1a	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	aefb9fc216	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	0a0b141278	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	9ba789d297	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	231f99b397	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	999b162786	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	fd5683f74b	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	1d6e7c6843	test: initialize capture structure before freeing ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	3 years ago
Naveen Albert	5199a70c07	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	2efcb5890e	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
Jaco Kroon	ef20afda63	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	086b1abf66	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
Naveen Albert	0bf6d7af33	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

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