asterisk

Commit Graph

Author	SHA1	Message	Date
Sperl Viktor	0bf31781c3	app_queue: indicate the paused state of a dynamically added member in queue_log. Fixes: #1021	5 months ago
George Joseph	92f361c826	Allow C++ source files (as extension .cc) in the main directory Although C++ files (as extension .cc) have been handled in the module directories for many years, the main directory was missing one line in its Makefile that prevented C++ files from being recognised there.	5 months ago
Naveen Albert	c157ee62f6	config.c: Make ast_variable_retrieve return last match. ast_variable_retrieve currently returns the first match for a variable, as opposed to the last one. This is problematic because modules that load config settings by explicitly calling ast_variable_retrieve on a variable name (as opposed to iterating through all the directives as specified) will end up taking the first specified value, such as the default value from the template rather than the actual effective value in an individual config section, leading to the wrong config. This fixes this by making ast_variable_retrieve return the last match, or the most recently overridden one, as the effective setting. This is similar to what the -1 index in the AST_CONFIG function does. There is another function, ast_variable_find_last_in_list, that does something similar. However, it's a slightly different API, and it sees virtually no usage in Asterisk. ast_variable_retrieve is what most things use so this is currently the relevant point of breakage. In practice, this is unlikely to cause any breakage, since there would be no logical reason to use an inherited value rather than an explicitly overridden value when loading a config. ASTERISK-30370 #close Resolves: #244 UpgradeNote: Config variables retrieved explicitly by name now return the most recently overriding value as opposed to the base value (e.g. from a template). This is equivalent to retrieving a config setting using the -1 index to the AST_CONFIG function. The major implication of this is that modules processing configs by explicitly retrieving variables by name will now get the effective value of a variable as overridden in a config rather than the first-set value (from a template), which is consistent with how other modules load config settings.	5 months ago
Alexey Khabulyak	634b57b8e6	format_gsm.c: Added mime type Sometimes it's impossible to get a file extension from URL (eg. http://example.com/gsm/your) so we have to rely on content-type header. Currenly, asterisk does not support content-type for gsm format(unlike wav). Added audio/gsm according to https://www.rfc-editor.org/rfc/rfc4856.html	5 months ago
Maksim Nesterov	8ec097a281	func_uuid: Add a new dialplan function to generate UUIDs This function is useful for uniquely identifying calls, recordings, and other entities in distributed environments, as well as for generating an argument for the AudioSocket application.	5 months ago
Sperl Viktor	d5c0486175	app_queue: allow dynamically adding a queue member in paused state. Fixes: #1007 UserNote: use the p option of AddQueueMember() for paused member state. Optionally, use the r(reason) option to specify a custom reason for the pause.	5 months ago
Naveen Albert	d09359b9d9	chan_iax2: Add log message for rejected calls. Add a log message for a path that currently silently drops IAX2 frames without indicating that anything is wrong.	5 months ago
Maximilian Fridrich	0f1f0ef3ef	chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams Currently, when a chan_pjsip channel receives a VIDUPDATE indication, an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is enabled on that endpoint. This restriction does not really make sense. Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as is the case with VP8, VP9, and H.265 streams). Resolves: #1013	5 months ago
Tinet-mucw	a23f28e7e0	audiohook.c: resolving the issue with audiohook both reading when packet loss on one side of the call When there is 0% packet loss on one side of the call and 15% packet loss on the other side, reading frame is often failed when reading direction_both audiohook. when read_factory available = 0, write_factory available = 320; i think write factory is usable read; because after reading one frame, there is still another frame that can be read together with the next read factory frame. Resolves: #851	5 months ago
Mike Pultz	b85e18b4ef	res_curl.conf.sample: clean up sample configuration and add new SSL options This update properly documents all the current configuration options supported by the curl implementation, including the new ssl_* options.	5 months ago
Viktor Litvinov	f30ad96b3f	res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big Set Mark bit in rtp stream when timestamp skew is bigger than MAX_TIMESTAMP_SKEW. Fixes: #927	5 months ago
Alexey Vasilyev	dc64d485d7	res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF Fixes #1004	5 months ago
Mike Pultz	e020ee5795	manager.c: Add Processed Call Count to CoreStatus output This update adds the processed call count to the CoreStatus AMI Action responsie. This output is similar to the values returned by "core show channels" or "core show calls" in the CLI. UserNote: The current processed call count is now returned as CoreProcessedCalls from the CoreStatus AMI Action.	5 months ago
Mike Pultz	3dae2cf88f	func_curl.c: Add additional CURL options for SSL requests This patch adds additional CURL TLS options / options to support mTLS authenticated requests: * ssl_verifyhost - perform a host verification on the peer certificate (CURLOPT_SSL_VERIFYHOST) * ssl_cainfo - define a CA certificate file (CURLOPT_CAINFO) * ssl_capath - define a CA certificate directory (CURLOPT_CAPATH) * ssl_cert - define a client certificate for the request (CURLOPT_SSLCERT) * ssl_certtype - specify the client certificate type (CURLOPT_SSLCERTTYPE) * ssl_key - define a client private key for the request (CURLOPT_SSLKEY) * ssl_keytype - specify the client private key type (CURLOPT_SSLKEYTYPE) * ssl_keypasswd - set a password for the private key, if required (CURLOPT_KEYPASSWD) UserNote: The following new configuration options are now available in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype', (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the libcurl documentation for more details.	5 months ago
Naveen Albert	f1c72347e9	sig_analog: Fix regression with FGD and E911 signaling. Commit `466eb4a52b` introduced a regression which completely broke Feature Group D and E911 signaling, by removing the call to analog_my_getsigstr, which affected multiple switch cases. Restore the original behavior for all protocols except Feature Group C CAMA (MF), which is all that patch was attempting to target. Resolves: #993	5 months ago
James Terhune	1aeecab6ae	main/stasis_channels.c: Fix crash when setting a global variable with invalid UTF8 characters Add check for null value of chan before referencing it with ast_channel_name() Resolves: #999	5 months ago
George Joseph	1646b78986	res_stir_shaken: Allow sending Identity headers for unknown TNs Added a new option "unknown_tn_attest_level" to allow Identity headers to be sent when a callerid TN isn't explicitly configured in stir_shaken.conf. Since there's no TN object, a private_key_file and public_cert_url must be configured in the attestation or profile objects. Since "unknown_tn_attest_level" uses the same enum as attest_level, some of the sorcery macros had to be refactored to allow sharing the enum and to/from string conversion functions. Also fixed a memory leak in crypto_utils:pem_file_cb(). Resolves: #921 UserNote: You can now set the "unknown_tn_attest_level" option in the attestation and/or profile objects in stir_shaken.conf to enable sending Identity headers for callerid TNs not explicitly configured.	5 months ago
George Joseph	2694f78d03	res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T The suppress_moh_on_sendonly endpoint option should have been defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES in the alembic script instead of OPT_YESNO_T and YESNO_VALUES. Also updated contrib/ast-db-manage/README.md to indicate that AST_BOOL_VALUES should always be used and provided an example. Resolves: #995	5 months ago
George Joseph	02e2000653	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.	5 months ago
Sean Bright	db79bcb82f	res_pjsip.c: Fix Contact header rendering for IPv6 addresses. Fix suggested by @nvsystems. Fixes #985	5 months ago
chrsmj	e9682be4ab	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	5 months ago
George Joseph	f8a0db8d15	func_pjsip_aor/contact: Fix documentation for contact ID Clarified the use of the contact ID returned from PJSIP_AOR. Resolves: #990	5 months ago
George Joseph	3e9b00a115	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	5 months ago
Thomas Guebels	5d26fef407	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	5 months ago
Naveen Albert	3cab4e7ab4	config.c Make ast_variable_update update last match. ast_variable_update currently sets the first match for a variable, as opposed to the last one. This issue is complementary to that raised in #244. This is incorrect and results in the wrong (or no) action being taken in cases where a section inherits from a template section. When the traversal occurs to update the setting, the existing code erroneously would use the first of possibly multiple matches in its update logic, which is wrong. Now, explicitly use the last match in the traversal, which will ensure that the actual setting is updated properly, and not skipped or ignored because a template from which the setting's section inherits was used for comparison. Resolves: #960 UpgradeNote: Config variables, when set/updated, such as via AMI, will now have the corresponding setting updated, even if their sections inherit from template sections.	5 months ago
Naveen Albert	7173c92d9f	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	5 months ago
George Joseph	d5a0626889	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.	6 months ago
Ben Ford	fbe8f8ff46	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.	6 months ago
Ben Ford	3457c7a6dc	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.	6 months ago
Thomas Guebels	ef6f3ef0e7	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	6 months ago
George Joseph	4fcd57f262	.github: Redirect NightlyAdmin to AsteriskNightlyAdmin	6 months ago
Naveen Albert	5782b038f0	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	6 months ago
George Joseph	7e079d5d22	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".	6 months ago
Sean Bright	c13ef65f21	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)`	6 months ago
George Joseph	35d6da06a6	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	6 months ago
Allan Nathanson	97770a947c	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	76601805b2	geolocation.sample.conf: Fix comment marker at end of file Resolves: #937	7 months ago
Sean Bright	b8ffe54e4b	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	06444290d0	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	d4358d5f7a	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	4623abb8bb	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	d856a3e2ef	alembic: Drop redundant voicemail_messages index. The `voicemail_messages_dir` index is a left prefix of the table's primary key and therefore unnecessary.	7 months ago
Sean Bright	e76f671810	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	7 months ago
Naveen Albert	d1bba7efc0	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	7 months ago
George Joseph	5d978ffcce	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	7 months ago
Ben Ford	e19da5d86f	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.	7 months ago
jiangxc	c0d3fbb5ae	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	7 months ago
Sean Bright	befe46162b	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.	7 months ago
Naveen Albert	a988506a3e	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	7 months ago
Peter Jannesen	adfc1846ef	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	7 months ago

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