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Author	SHA1	Message	Date
Ben Ford	b8d818bd3c	Add res_pjsip_config_sangoma external module. ... Adds res_pjsip_config_sangoma as an external module that can be downloaded via menuselect. It lives under the Resource Modules section.	7 months ago
Ben Ford	8273eefd87	app_mixmonitor: Add 'D' option for dual-channel audio. ... Adds the 'D' option to app_mixmonitor that interleaves the input and output frames of the channel being recorded in the monitor output frame. This allows for two streams in the recording: the transmitted audio and the received audio. The 't' and 'r' options are compatible with this. Fixes: #945 UserNote: The MixMonitor application now has a new 'D' option which interleaves the recorded audio in the output frames. This allows for stereo recording output with one channel being the transmitted audio and the other being the received audio. The 't' and 't' options are compatible with this.	7 months ago
Thomas Guebels	6763dda90f	pjsip_transport_events: Avoid monitor destruction ... When a transport is disconnected, several events can arrive following each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it will trigger the destruction of the transport monitor object. The lookup for the transport monitor to destroy is done using the transport key, that contains the transport destination host:port. A reconnect attempt by pjsip will be triggered as soon something needs to send a packet using that transport. This can happen directly after a disconnect since ca Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will also try to trigger the destruction of the transport monitor if not already done. Since the lookup for the transport monitor to destroy is done using the transport key, it can match newly created transports towards the same destination and destroy their monitor object. Because of this, it was sometimes not possible to monitor a transport after one or more disconnections. This fix adds an additional check on the transport pointer to ensure only a monitor for that specific transport is removed. Fixes: #923	7 months ago
George Joseph	3b2bde6c48	.github: Redirect NightlyAdmin to AsteriskNightlyAdmin	7 months ago
George Joseph	ce322ad16a	.github: NightlyAdmin now calls external CloseStaleIssuesAndPRs	7 months ago
Naveen Albert	6a07e8e0f1	app_dial: Fix progress timeout calculation with no answer timeout. ... If to_answer is -1, simply comparing to see if the progress timeout is smaller than the answer timeout to prefer it will fail. Add an additional check that chooses the progress timeout if there is no answer timeout (or as before, if the progress timeout is smaller). Resolves: #821	7 months ago
George Joseph	23218032ef	pjproject_bundled: Tweaks to support out-of-tree development ... * pjproject is now configured with --disable-libsrtp so it will build correctly when doing "out-of-tree" development. Asterisk doesn't use pjproject for handling media so pjproject doesn't need libsrtp itself. * The pjsua app (which we used to use for the testsuite) no longer builds in pjproject's master branch so we just skip it. The testsuite no longer needs it anyway. See third-party/pjproject/README-hacking.md for more info on building pjproject "out-of-tree".	7 months ago
Sean Bright	f3138af519	Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets." ... This reverts commit cb5e3445be. The original change from 16 to 15 bit sequence numbers was predicated on the following from the now-defunct libSRTP FAQ on sourceforge.net: > *Q6. The use of implicit synchronization via ROC seems > dangerous. Can senders and receivers lose ROC synchronization?* > > **A.** It is possible to lose ROC synchronization between sender and > receiver(s), though it is not likely in practice, and practical > steps can be taken to avoid it. A burst loss of 2^16 packets or more > will always break synchronization. For example, a conversational > voice codec that sends 50 packets per second will have its ROC > increment about every 22 minutes. A network with a burst of packet > loss that long has problems other than ROC synchronization. > > There is a higher sensitivity to loss at the very outset of an SRTP > stream. If the sender's initial sequence number is close to the > maximum value of 2^16-1, and all packets are lost from the initial > packet until the sequence number cycles back to zero, the sender > will increment its ROC, but the receiver will not. The receiver > cannot determine that the initial packets were lost and that > sequence-number rollover has occurred. In this case, the receiver's > ROC would be zero whereas the sender's ROC would be one, while their > sequence numbers would be so close that the ROC-guessing algorithm > could not detect this fact. > > There is a simple solution to this problem: the SRTP sender should > randomly select an initial sequence number that is always less than > 2^15. This ensures correct SRTP operation so long as fewer than 2^15 > initial packets are lost in succession, which is within the maximum > tolerance of SRTP packet-index determination (see Appendix A and > page 14, first paragraph of RFC 3711). An SRTP receiver should > carefully implement the index-guessing algorithm. A naive > implementation can unintentionally guess the value of > 0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15 > and the locally stored SEQ and ROC are zero. (This can happen when > the implementation fails to treat those zero values as a special > case.) > > When ROC synchronization is lost, the receiver will not be able to > properly process the packets. If anti-replay protection is turned > on, then the desynchronization will appear as a burst of replay > check failures. Otherwise, if authentication is being checked, then > it will appear as a burst of authentication failures. Otherwise, if > encryption is being used, the desynchronization may not be detected > by the SRTP layer, and the packets may be improperly decrypted. However, modern libSRTP (as of 1.0.1[1]) now mentions the following in their README.md[2]: > The sequence number in the rtp packet is used as the low 16 bits of > the sender's local packet index. Note that RTP will start its > sequence number in a random place, and the SRTP layer just jumps > forward to that number at its first invocation. An earlier version > of this library used initial sequence numbers that are less than > 32,768; this trick is no longer required as the > rdbx_estimate_index(...) function has been made smarter. So truncating our initial sequence number to 15 bit is no longer necessary. 1. 0eb007f0dc/CHANGES (L271-L289) 2. 2de20dd9e9/README.md (implementation-notes)	7 months ago
George Joseph	e434203810	core_unreal.c: Fix memory leak in ast_unreal_new_channels() ... When the channel tech is multistream capable, the reference to chan_topology was passed to the new channel. When the channel tech isn't multistream capable, the reference to chan_topology was never released. "Local" channels are multistream capable so it didn't affect them but the confbridge "CBAnn" and the bridge_media "Recorder" channels are not so they caused a leak every time one of them was created. Also added tracing to ast_stream_topology_alloc() and stream_topology_destroy() to assist with debugging. Resolves: #938	7 months ago
Allan Nathanson	4a3319a587	dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin ... The dnsmgr_refresh() function checks to see if the IP address associated with a name/service has changed. The gotcha is that the ast_get_ip_or_srv() function only returns the first IP address returned by the DNS query. If there are multiple IPs associated with the name and the returned order is not consistent (e.g. with DNS round-robin) then the other IP addresses are not included in the comparison and the entry is flagged as changed even though the IP is still valid. Updated the code to check all IP addresses and flag a change only if the original IP is no longer valid. Resolves: #924	7 months ago
George Joseph	00d72b0e9e	geolocation.sample.conf: Fix comment marker at end of file ... Resolves: #937	7 months ago
Sean Bright	7c359a2dd3	func_base64.c: Ensure we set aside enough room for base64 encoded data. ... Reported by SingularTricycle on IRC. Fixes #940	7 months ago
Naveen Albert	97dfe4cd40	app_dial: Fix progress timeout. ... Under some circumstances, the progress timeout feature added in commit 320c98eec8 does not work as expected, such as if there is no media flowing. Adjust the waitfor call to explicitly use the progress timeout if it would be reached sooner than the answer timeout to ensure we handle the timers properly. Resolves: #821	7 months ago
Naveen Albert	63e9cc19d1	chan_dahdi: Never send MWI while off-hook. ... In some circumstances, it is possible for the do_monitor thread to erroneously think that a line is on-hook and send an MWI FSK spill to it when the line is really off-hook and no MWI should be sent. Commit 0a8b3d3467 previously fixed this issue in a more readily encountered scenario, but it has still been possible for MWI to be sent when it shouldn't be. To robustly fix this issue, query DAHDI for the hook status to ensure we don't send MWI on a line that is actually still off hook. Resolves: #928	7 months ago
George Joseph	5cf699370b	manager.c: Add unit test for Originate app and appdata permissions ... This unit test checks that dialplan apps and app data specified as parameters for the Originate action are allowed with the permissions the user has.	7 months ago
Sean Bright	f97ad807ff	alembic: Drop redundant voicemail_messages index. ... The `voicemail_messages_dir` index is a left prefix of the table's primary key and therefore unnecessary.	8 months ago
Sean Bright	243f20a78d	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. ... Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922	8 months ago
Naveen Albert	b8b21b3f00	main, res, tests: Fix compilation errors on FreeBSD. ... asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings. config.c: Include header file for WIFEXITED/WEXITSTATUS macros. res_timing_kqueue: Use more portable format specifier. test_crypto: Use non-linux limits.h header file. Resolves: #916	8 months ago
George Joseph	7db8ae296c	res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs ... In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned success but with a timeout of 0, we were stopping the timer and decrementing the refcount on instance but not resetting the timeout_timer to -1. When dtls_srtp_stop_timeout_timer() was later called, it was atempting to stop a stale timer and could decrement the refcount on instance again which would then cause the instance destructor to run early. This would result in either a FRACK or a SEGV when ast_rtp_stop(0 was called. According to the OpenSSL docs, we shouldn't have been stopping the timer when DTLSv1_get_timeout() returned success and the new timeout was 0 anyway. We should have been calling DTLSv1_handle_timeout() again immediately so we now reschedule the timer callback for 1ms (almost immediately). Additionally, instead of scheduling the timer callback at a fixed interval returned by the initial call to DTLSv1_get_timeout() (usually 999 ms), we now reschedule the next callback based on the last call to DTLSv1_get_timeout(). Resolves: #487	8 months ago
Ben Ford	e8d3869e75	manager.c: Restrict ModuleLoad to the configured modules directory. ... When using the ModuleLoad AMI action, it was possible to traverse upwards through the directories to files outside of the configured modules directory. We decided it would be best to restrict access to modules exclusively in the configured directory. You will now get an error when the specified module is outside of this limitation. Fixes: #897 UserNote: The ModuleLoad AMI action now restricts modules to the configured modules directory.	8 months ago
jiangxc	2d676c7560	res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE` ... When using the speech recognition module, crashes can occur sporadically due to a "double free or corruption (out)" error. Now, in the section where the audio stream is being captured in a loop, each time after releasing fr, it is set to NULL to prevent repeated deallocation. Fixes #772	8 months ago
Sean Bright	084c04f711	cdr_custom: Allow absolute filenames. ... A follow up to #893 that brings the same functionality to cdr_custom. Also update the sample configuration files to note support for absolute paths.	8 months ago
Naveen Albert	7c982de5c6	astfd.c: Avoid calling fclose with NULL argument. ... Don't pass through a NULL argument to fclose, which is undefined behavior, and instead return -1 and set errno appropriately. This also avoids a compiler warning with glibc 2.38 and newer, as glibc commit 71d9e0fe766a3c22a730995b9d024960970670af added the nonnull attribute to this argument. Resolves: #900	8 months ago
Peter Jannesen	f3e74d34ce	channel: Preserve CHANNEL(userfield) on masquerade. ... In certain circumstances a channel may undergo an operation referred to as a masquerade. If this occurs the CHANNEL(userfield) value was not preserved causing it to get lost. This change makes it so that this field is now preserved. Fixes: #882	8 months ago
Peter Jannesen	0ed7f5800d	cel_custom: Allow absolute filenames. ... If a filename starts with a '/' in cel_custom [mappings] assume it is a absolute file path and not relative filename/path to AST_LOG_DIR/cel_custom/	8 months ago
Naveen Albert	9423710a5e	app_voicemail: Fix ill-formatted pager emails with custom subject. ... Add missing end-of-headers newline to pager emails with custom subjects, since this was missing from this code path. Resolves: #902	8 months ago
Sean Bright	3243316dac	res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery ... Fixes #895	8 months ago
George Joseph	8155b0e50f	stir_shaken: Fix propagation of attest_level and a few other values ... attest_level, send_mky and check_tn_cert_public_url weren't propagating correctly from the attestation object to the profile and tn. * In the case of attest_level, the enum needed to be changed so the "0" value (the default) was "NOT_SET" instead of "A". This now allows the merging of the attestation object, profile and tn to detect when a value isn't set and use the higher level value. * For send_mky and check_tn_cert_public_url, the tn default was forced to "NO" which always overrode the profile and attestation objects. Their defaults are now "NOT_SET" so the propagation happens correctly. * Just to remove some redundant code in tn_config.c, a bunch of calls to generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were replaced with a single call to generate_acfg_common_sorcery_handlers(). Resolves: #904	8 months ago
George Joseph	48acf00eab	Fix application references to Background ... The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages.	8 months ago
George Joseph	ce35f5684b	manager.conf.sample: Fix mathcing typo	8 months ago
George Joseph	92c0bd4b50	manager: Enhance event filtering for performance ... UserNote: You can now perform more granular filtering on events in manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/` This is much more efficient than `eventfilter = Event: Newchannel.*Channel: PJSIP/` Full syntax guide is in configs/samples/manager.conf.sample.	8 months ago
George Joseph	a26e79dfe2	manager.c: Split XML documentation to manager_doc.xml	8 months ago
George Joseph	94237f684a	.github: Fix realtime param on Weekly and Nightly tests and... ... Rename the "Cleanup" job in the cherry-pick and recheck jobs to "Summary".	8 months ago
George Joseph	60ad3efc9d	.github: Add WeeklyTests and make Nightlies Monday-Saturday ... ...and add "realtime" option.	8 months ago
George Joseph	cc06c95ed2	db.c: Remove limit on family/key length ... Consumers like media_cache have been running into issues with the previous astdb "/family/key" limit of 253 bytes when needing to store things like long URIs. An Amazon S3 URI is a good example of this. Now, instead of using a static 256 byte buffer for "/family/key", we use ast_asprintf() to dynamically create it. Both test_db.c and test_media_cache.c were also updated to use keys/URIs over the old 253 character limit. Resolves: #881 UserNote: The `ast_db_*()` APIs have had the 253 byte limit on "/family/key" removed and will now accept families and keys with a total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*` dialplan applications, dialplan functions, manager actions and `databse` CLI commands. Since the media_cache also uses the `ast_db_*()` APIs, you can now store resources with URIs longer than 253 bytes.	8 months ago
George Joseph	24d641f40a	.github: Changes required to use cached builds and shorten names	8 months ago
George Joseph	c33ff71708	res_stir_shaken: Remove stale include for jansson.h in verification.c ... verification.c had an include for jansson.h left over from previous versions of the module. Since res_stir_shaken no longer has a dependency on jansson, the bundled version wasn't added to GCC's include path so if you didn't also have a jansson development package installed, the compile would fail. Removing the stale include was the only thing needed. Resolves: #889	8 months ago
George Joseph	03d0d1eba4	res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid ... * If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID, check_for_old_config() now returns LOAD_DECLINE instead of continuing on with a bad pointer. * If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config() assumes the config is being loaded from realtime and now returns LOAD_SUCCESS. If it's actually not being loaded from realtime, sorcery will catch that later on. * Also refactored the error handling in load_module() a bit. Resolves: #884	8 months ago
George Joseph	7773327546	res_stir_shaken: Check for disabled before param validation ... For both attestation and verification, we now check whether they've been disabled either globally or by the profile before validating things like callerid, orig_tn, dest_tn, etc. This prevents useless error messages. Resolves: #879	8 months ago
Tinet-mucw	a721f99eb0	app_chanspy.c: resolving the issue writing frame to whisper audiohook. ... ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook. Resolves: #876	8 months ago
Alexei Gradinari	b9b0cffc29	autoservice: Do not sleep if autoservice_stop is called within autoservice thread ... It's possible that ast_autoservice_stop is called within the autoservice thread. In this case the autoservice thread is stuck in an endless sleep. To avoid endless sleep ast_autoservice_stop must check that it's not called within the autoservice thread. Fixes: #763	8 months ago
George Joseph	3265d66e98	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
George Joseph	d6157aff5b	app_voicemail: Use ast_asprintf to create mailbox SQL query ... ...instead of trying to calculate the length of the buffer needed manually.	9 months ago
Mike Bradeen	ac673dd14e	res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch ... When Asterisk sends an offer to Bob that includes 48K and 8K codecs with matching 4733 offers, Bob may want to use the 48K audio codec but can not accept 48K digits and so negotiates for a mixed set. Asterisk will now check Bob's offer to make sure Bob has indicated this is acceptible and if not, will use Bob's preference. Fixes: #847	9 months ago
Tinet-mucw	e98127d540	app_chanspy.c: resolving the issue with audiohook direction read ... ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail. Resolves: #861	9 months ago
George Joseph	ca60f7db8f	security_agreements.c: Refactor the to_str functions and fix a few other bugs ... * A static array of security mechanism type names was created. * ast_sip_str_to_security_mechanism_type() was refactored to do a lookup in the new array instead of using fixed "if/else if" statments. * security_mechanism_to_str() and ast_sip_security_mechanisms_to_str() were refactored to use ast_str instead of a fixed length buffer to store the result. * ast_sip_security_mechanism_type_to_str was removed in favor of just referencing the new type name array. Despite starting with "ast_sip_", it was a static function so removing it doesn't affect ABI. * Speaking of "ast_sip_", several other static functions that started with "ast_sip_" were renamed to avoid confusion about their public availability. * A few VECTOR free loops were replaced with AST_VECTOR_RESET(). * Fixed a meomry leak in pjsip_configuration.c endpoint_destructor caused by not calling ast_sip_security_mechanisms_vector_destroy(). * Fixed a memory leak in res_pjsip_outbound_registration.c add_security_headers() caused by not specifying OBJ_NODATA in an ao2_callback. * Fixed a few ao2_callback return code misuses. Resolves: #845	9 months ago
Alexei Gradinari	95fadcf6db	res_pjsip_sdp_rtp fix leaking astobj2 ast_format ... PR #700 added a preferred_format for the struct ast_rtp_codecs, but when set the preferred_format it leaks an astobj2 ast_format. In the next code ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0)); both functions ast_rtp_codecs_set_preferred_format and ast_format_cap_get_format increases the ao2 reference count. Fixes: #856	9 months ago
George Joseph	7b0478f17f	stir_shaken.conf.sample: Fix bad references to private_key_path ... They should be private_key_file. Resolves: #854	9 months ago
Sean Bright	ab9c476dd2	res_pjsip_logger.c: Fix 'OPTIONS' tab completion. ... Fixes #843	9 months ago
Sean Bright	c5fbf5dcae	alembic: Make 'revises' header comment match reality.	9 months ago