asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	badaf0e383	res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party. Resolves: #979 UserNote: The new "suppress_moh_on_sendonly" endpoint option can be used to prevent playing MOH back to a caller if the remote end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP. (cherry picked from commit `03ab004e26`)	12 months ago
Sean Bright	94fcb3174b	res_pjsip.c: Fix Contact header rendering for IPv6 addresses. Fix suggested by @nvsystems. Fixes #985 (cherry picked from commit `1c0fcd583d`)	12 months ago
chrsmj	6d98127a75	samples: remove and/or change some wiki mentions Cleaned some dead links. Replaced word wiki with either docs or link to https://docs.asterisk.org/ Resolves: #974 (cherry picked from commit `c037fc7fae`)	12 months ago
George Joseph	057de6ab66	func_pjsip_aor/contact: Fix documentation for contact ID Clarified the use of the contact ID returned from PJSIP_AOR. Resolves: #990 (cherry picked from commit `aa736dac12`)	12 months ago
George Joseph	2a48183169	res_pjsip: Move tenantid to end of ast_sip_endpoint The tenantid field was originally added to the ast_sip_endpoint structure at the end of the AST_DECLARE_STRING_FIELDS block. This caused everything after it in the structure to move down in memory and break ABI compatibility. It's now at the end of the structure as an AST_STRING_FIELD_EXTENDED. Given the number of string fields in the structure now, the initial string field allocation was also increased from 64 to 128 bytes. Resolves: #982 (cherry picked from commit `144033db31`)	12 months ago
Thomas Guebels	6795ded3f4	pjsip_transport_events: handle multiple addresses for a domain The key used for transport monitors was the remote host name for the transport and not the remote address resolved for this domain. This was problematic for domains returning multiple addresses as several transport monitors were created with the same key. Whenever a subsystem wanted to register a callback it would always end up attached to the first transport monitor with a matching key. The key used for transport monitors is now the remote address and port the transport actually connected to. Fixes: #932 (cherry picked from commit `184a30efac`)	12 months ago
Naveen Albert	07166788fd	func_evalexten: Add EVAL_SUB function. This adds an EVAL_SUB function, which is similar to the existing EVAL_EXTEN function but significantly more powerful, as it allows executing arbitrary dialplan and capturing its return value as the function's output. While EVAL_EXTEN should be preferred if it is possible to use it, EVAL_SUB can be used in a wider variety of cases and allows arbitrary computation to be performed in a dialplan function call, leveraging the dialplan. Resolves: #951 (cherry picked from commit `34662cf0ea`)	12 months ago
George Joseph	0e5c832703	res_srtp: Change Unsupported crypto suite msg from verbose to debug There's really no point in spamming logs with a verbose message for every unsupported crypto suite an older client may send in an SDP. If none are supported, there will be an error or warning. (cherry picked from commit `2717068e1b`)	12 months ago
Ben Ford	e6822d7647	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. (cherry picked from commit `9f1f3703c9`)	12 months ago
Ben Ford	f9278c1790	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. (cherry picked from commit `4616408b09`)	12 months ago
Thomas Guebels	db312ab0ef	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 (cherry picked from commit `008ad8098a`)	12 months ago
George Joseph	f93b97ac79	.github: Redirect NightlyAdmin to AsteriskNightlyAdmin (cherry picked from commit `32193addcc`)	12 months ago
Naveen Albert	33e8398e2b	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 (cherry picked from commit `4dd6074f9f`)	12 months ago
George Joseph	0f3d8f21c8	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". (cherry picked from commit `142bdeff5b`)	12 months ago
Sean Bright	76d1ef4d2f	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)` (cherry picked from commit `a39384721d`)	12 months ago
George Joseph	2f8d008103	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 (cherry picked from commit `41162f963b`)	12 months ago
Allan Nathanson	039baa0317	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 (cherry picked from commit `338e191539`)	12 months ago
George Joseph	2425d2218c	geolocation.sample.conf: Fix comment marker at end of file Resolves: #937 (cherry picked from commit `d92af8acaf`)	12 months ago
Sean Bright	166a309549	func_base64.c: Ensure we set aside enough room for base64 encoded data. Reported by SingularTricycle on IRC. Fixes #940 (cherry picked from commit `349db1de3d`)	12 months ago
Naveen Albert	556b974d1b	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 (cherry picked from commit `54f45c6b86`)	12 months ago
Naveen Albert	43166eb85b	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 (cherry picked from commit `4635a5ff4d`)	12 months ago
George Joseph	04b882397f	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. (cherry picked from commit `9ecccb21ed`)	12 months ago
Sean Bright	0bba91a49d	alembic: Drop redundant voicemail_messages index. The `voicemail_messages_dir` index is a left prefix of the table's primary key and therefore unnecessary. (cherry picked from commit `0bdfebd1b7`)	12 months ago
Sean Bright	682ad186fb	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 (cherry picked from commit `e1f2866bb6`)	12 months ago
Naveen Albert	ab79321a29	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 (cherry picked from commit `b939d840db`)	12 months ago
George Joseph	cbbd2a6b47	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 (cherry picked from commit `fe5bea34ab`)	12 months ago
Ben Ford	c7beb10006	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. (cherry picked from commit `ee9a0f056e`)	12 months ago
jiangxc	af05288825	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 (cherry picked from commit `f6d1dfd999`)	12 months ago
Sean Bright	65fadcbc09	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. (cherry picked from commit `2618a4f1e5`)	12 months ago
Naveen Albert	1372eeba51	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 (cherry picked from commit `2c583be963`)	12 months ago
Peter Jannesen	2f44ac22b2	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 (cherry picked from commit `fd196e9113`)	12 months ago
Peter Jannesen	d9e01cf0a1	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/ (cherry picked from commit `b7c4be9d5b`)	12 months ago
Naveen Albert	e0edea0383	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 (cherry picked from commit `e63da10a17`)	12 months ago
Sean Bright	4ade5a21b7	res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery Fixes #895 (cherry picked from commit `dad84444fc`)	12 months ago
George Joseph	6dc00f642a	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. (cherry picked from commit `f71668f0b9`)	12 months ago
George Joseph	82a229166a	manager.conf.sample: Fix mathcing typo (cherry picked from commit `adaacc281d`)	12 months ago
George Joseph	56d3070939	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. (cherry picked from commit `b8d16fd800`)	12 months ago
George Joseph	2ca93f0cdc	manager.c: Split XML documentation to manager_doc.xml (cherry picked from commit `dd643bf827`)	12 months ago
George Joseph	03883ccb8e	.github: Fix realtime param on Weekly and Nightly tests and... Rename the "Cleanup" job in the cherry-pick and recheck jobs to "Summary". (cherry picked from commit `0c5a114b9f`)	12 months ago
George Joseph	5e9ea95517	.github: Add WeeklyTests and make Nightlies Monday-Saturday ...and add "realtime" option. (cherry picked from commit `d17bf1af1a`)	12 months ago
George Joseph	6297462fd2	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. (cherry picked from commit `ff7d094803`)	12 months ago
George Joseph	7788fac511	.github: Changes required to use cached builds and shorten names (cherry picked from commit `6c0fa2274b`)	12 months ago
Asterisk Development Team	8e4a09f711	Update for 22.0.0	1 year ago
Asterisk Development Team	4473ed5256	Update for 22.0.0-rc2	1 year ago
George Joseph	fd8d28cb60	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	1 year ago
George Joseph	dc8f83662b	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	1 year ago
George Joseph	4411f22d67	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	1 year ago
Asterisk Development Team	7c1687ba12	Update for 22.0.0-rc1	1 year ago
George Joseph	3fb205f3af	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 (cherry picked from commit `ef7d177ee3`)	1 year ago
Tinet-mucw	ef2bae8464	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 (cherry picked from commit `ffb1dca485`)	1 year ago

1 2 3 4 5 ...

33985 Commits (badaf0e38399a30fd7409eb7f0b2f9658390a471) All Branches Search

33985 Commits (badaf0e38399a30fd7409eb7f0b2f9658390a471)

All Branches