asterisk

Commit Graph

Author	SHA1	Message	Date
Kent	0fbb6de6f4	res_pjsip: Add new AOR option "qualify_2xx_only" Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify feature to mark a contact as available only if an OPTIONS request returns a 2XX response. If the option is not specified or is false, any response to the OPTIONS request marks the contact as available. UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only" option that can be set so that only 2XX responses to OPTIONS requests used to qualify a contact will mark the contact as available. (cherry picked from commit `21dba60ff2`)	11 months ago
George Joseph	68359a3f96	.github: Change the run name for OnPRStateChangedPriv (cherry picked from commit `865c8b8232`)	11 months ago
Jaco Kroon	20cf8b3a07	res_odbc: release threads from potential starvation. Whenever a slot is freed up due to a failed connection, wake up a waiter before failing. In the case of a dead connection there could be waiters, for example, let's say two threads tries to acquire objects at the same time, with one in the cached connections, one will acquire the dead connection, and the other will enter into the wait state. The thread with the dead connection will clear up the dead connection, and then attempt a re-acquire (at this point there cannot be cached connections else the other thread would have received that and tried to clean up), as such, at this point we're guaranteed that either there are no waiting threads, or that the maxconnections - connection_cnt threads will attempt to re-acquire connections, and then either succeed, using those connections, or failing, and then signalling to release more waiters. Also fix the pointer log for ODBC handle %p dead which would always reflect NULL. Signed-off-by: Jaco Kroon <jaco@uls.co.za> (cherry picked from commit `9b45103bc8`)	11 months ago
Sperl Viktor	02bba40fca	app_queue: indicate the paused state of a dynamically added member in queue_log. Fixes: #1021 (cherry picked from commit `d832aae1d7`)	11 months ago
George Joseph	0a5a6995af	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. (cherry picked from commit `9a1180c6af`)	11 months ago
Alexey Khabulyak	b26b73790f	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 (cherry picked from commit `ced3b41327`)	11 months ago
Maksim Nesterov	186e9b5bb8	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. (cherry picked from commit `c50dea93a5`)	11 months ago
Sperl Viktor	94a32885e7	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. (cherry picked from commit `7b7df5d30e`)	11 months ago
Naveen Albert	6f592e7163	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. (cherry picked from commit `be2bea854e`)	11 months ago
Maximilian Fridrich	0bd4d3320a	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 (cherry picked from commit `45c5d39c58`)	11 months ago
Tinet-mucw	1a71ea7456	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 (cherry picked from commit `3ac5f82d49`)	11 months ago
Mike Pultz	a8d81692dd	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. (cherry picked from commit `424e4ee761`)	11 months ago
Viktor Litvinov	ada5a706b7	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 (cherry picked from commit `2599aa3be0`)	11 months ago
Alexey Vasilyev	7d681cbd90	res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF Fixes #1004 (cherry picked from commit `e96d1cf2ae`)	11 months ago
Mike Pultz	2d4460ad4d	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. (cherry picked from commit `0c193b725b`)	11 months ago
Mike Pultz	6b194cbbe3	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. (cherry picked from commit `8e289af9de`)	11 months ago
Naveen Albert	42629c695d	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 (cherry picked from commit `4f0a506c67`)	11 months ago
James Terhune	fc390cf785	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 (cherry picked from commit `d6b7554012`)	11 months ago
George Joseph	7a640ca8a8	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. (cherry picked from commit `e9f336b47b`)	11 months ago
Asterisk Development Team	8d1be28001	Update for 22.1.1	12 months ago
Ben Ford	4d80d7ab22	manager.c: Restrict ListCategories to the configuration directory. When using the ListCategories AMI action, it was possible to traverse upwards through the directories to files outside of the configured configuration directory. This action is now restricted to the configured directory and an error will now be returned if the specified file is outside of this limitation. Resolves: #GHSA-33x6-fj46-6rfh UserNote: The ListCategories AMI action now restricts files to the configured configuration directory.	12 months ago
Asterisk Development Team	a66c9decfe	Update for 22.1.0	1 year ago
George Joseph	95e3e202cc	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	1 year ago
Asterisk Development Team	65d531c8e9	Update for 22.1.0-rc1	1 year ago
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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago
George Joseph	f93b97ac79	.github: Redirect NightlyAdmin to AsteriskNightlyAdmin (cherry picked from commit `32193addcc`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago
George Joseph	2425d2218c	geolocation.sample.conf: Fix comment marker at end of file Resolves: #937 (cherry picked from commit `d92af8acaf`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago

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