asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	e4de46e847	bridge_softmix: Suppress error on topology change failure There are conditions under which a failure to change topology is expected so there's no need to print an ERROR message. ASTERISK-29618 Reported by: Alexander Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481	4 years ago
sungtae kim	f9b8a3a8c8	resource_channels.c: Fix wrong external media parameter parse Fixed ARI external media handler to accept body parameters. ASTERISK-29622 Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d	4 years ago
Sean Bright	1bd642db7f	config_options: Handle ACO arrays correctly in generated XML docs. There are 3 separate changes here but they are all closely related: * Only try to set matchfield attributes on 'field' nodes * We need to adjust how we treat the category pointer based on the value of the category_match, to avoid memory corruption. We now generate a regex-like string when match types other than ACO_WHITELIST and ACO_BLACKLIST are used. * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to ACO_BLACKLIST_EXACT since we only have one category we need to ignore, not two. ASTERISK-29614 #close Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e	4 years ago
Naveen Albert	38cc3160d3	chan_iax2: Add ANI2/OLI information element Adds an information element for ANI2 so that Originating Line Information can be transmitted over IAX2 channels. ASTERISK-29605 #close Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2	4 years ago
Mark Murawski	efcde5c582	pbx_ael: Fix crash and lockup issue regarding 'ael reload' Currently pbx_ael does not check if a reload is currently pending before proceeding with a reload. This can cause multiple threads to operate at the same time on what should be mutex protected data. This change adds protection to reloading to ensure only one ael reload is executing at a time. ASTERISK-29609 #close Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35	4 years ago
Naveen Albert	c5c5171ec8	app_read: Allow reading # as a digit Allows for the digit # to be read as a digit, just like any other DTMF digit, as opposed to forcing it to be used as an end of input indicator. The default behavior remains unchanged. ASTERISK-18454 #close Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b	4 years ago
Sebastien Duthil	166556961b	res_rtp_asterisk: Automatically refresh stunaddr from DNS This allows the STUN server to change its IP address without having to reload the res_rtp_asterisk module. The refresh of the name resolution occurs first when the module is loaded, then recurringly, slightly after the previous DNS answer TTL expires. ASTERISK-29508 #close Change-Id: I7955a046293f913ba121bbd82153b04439e3465f	4 years ago
Naveen Albert	220ce865db	app_queue: Don't reset queue stats on reload Prevents reloads of app_queue from also resetting queue statistics. Also preserves individual queue agent statistics if we're just reloading members. ASTERISK-28701 Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1	4 years ago
Naveen Albert	398a686fac	bridge_basic: Change warning to verbose if transfer cancelled The attended transfer feature will emit a warning if the user cancels the transfer or the attended transfer doesn't complete for any reason. Changes the warning to a verbose message, since nothing is actually wrong here. ASTERISK-29612 #close Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d	4 years ago
Alexander Traud	f516c57401	res_rtp_asterisk: sqrt(.) requires the header math.h. ASTERISK-29616 Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900	4 years ago
Alexander Traud	e2fe0a049a	dialplan: Add one static and fix two whitespace errors. Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83	4 years ago
Sarah Autumn	b731b810b9	sig_analog: Changes to improve electromechanical signalling compatibility This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	4 years ago
Andre Barbosa	f57fc18657	media_cache: Don't lock when curl the remote file When playing a remote sound file, which is not in cache, first we need to download it with ast_bucket_file_retrieve. This can take a while if the remote host is slow. The current CURL timeout is 180secs, so in extreme situations, it can take 3 minutes to return. Because ast_media_cache_retrieve has a lock on all function, while we are waiting for the delayed download, Asterisk is not able to play any more files, even the files already cached locally. ASTERISK-29544 #close Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408	4 years ago
George Joseph	18c467686c	res_pjproject: Allow mapping to Asterisk TRACE level Allow mapping pjproject log messages to the Asterisk TRACE log level. The defaults were also changes to log pjproject levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6 all went to DEBUG. ASTERISK-29582 Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d	4 years ago
Naveen Albert	9ee7b884f1	app_milliwatt: Timing fix The Milliwatt application uses incorrect tone timings that cause it to play the 1004 Hz tone constantly. This adds an option to enable the correct timing behavior, so that the Milliwatt application can be used for milliwatt test lines. The default behavior remains unchanged for compatability reasons, even though it is incorrect. ASTERISK-29575 #close Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c	4 years ago
Naveen Albert	7f4433f8c6	app_morsecode: Add American Morse code Previously, the Morsecode application only supported international Morse code. This adds support for American Morse code and adds an option to configure the frequency used in off intervals. Additionally, the application checks for hangup between tones to prevent application execution from continuing after hangup. ASTERISK-29541 Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4	4 years ago
Naveen Albert	4bd5757123	func_math: Return integer instead of float if possible The MIN, MAX, and ABS functions all support float arguments, but currently return floats even if the arguments are all integers and the response is a whole number, in which case the user is likely expecting an integer. This casts the float to an integer before printing into the response buffer if possible. ASTERISK-29495 Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488	4 years ago
Naveen Albert	1055660252	func_scramble: Audio scrambler function Adds a function to scramble audio on a channel using whole spectrum frequency inversion. This can be used as a privacy enhancement with applications like ChanSpy or other potentially sensitive audio. ASTERISK-29542 Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e	4 years ago
Naveen Albert	4606fb604a	app_originate: Add ability to set codecs A list of codecs to use for dialplan-originated calls can now be specified in Originate, similar to the ability in call files and the manager action. Additionally, we now default to just using the slin codec for originated calls, rather than all the slin* codecs up through slin192, which has been known to cause issues and inconsistencies from AMI and call file behavior. ASTERISK-29543 Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883	4 years ago
Alexander Traud	478c4fa588	BuildSystem: Remove two dead exceptions for compiler Clang. Commit `305ce3d` added -Wno-parentheses-equality to Makefile.rules, turning the previous two warning suppressions from commit `e9520db` redundant. Let us remove the latter. Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac	4 years ago
Sean Bright	b02538de2e	mgcp: Remove dead debug code ASTERISK-20339 #close Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5	4 years ago
Joshua C. Colp	68adc2304f	policy: Add deprecation and removal versions to modules. app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. cdr_mysql was deprecated in 1.8, to be removed in 19. app_mysql was deprecated in 1.8, to be removed in 19. app_ices was deprecated in 16, to be removed in 19. app_macro was deprecated in 16, to be removed in 21. app_fax was deprecated in 16, to be removed in 19. app_url was deprecated in 16, to be removed in 19. app_image was deprecated in 16, to be removed in 19. app_nbscat was deprecated in 16, to be removed in 19. app_dahdiras was deprecated in 16, to be removed in 19. cdr_syslog was deprecated in 16, to be removed in 19. chan_oss was deprecated in 16, to be removed in 19. chan_phone was deprecated in 16, to be removed in 19. chan_sip was deprecated in 17, to be removed in 21. chan_nbs was deprecated in 16, to be removed in 19. chan_misdn was deprecated in 16, to be removed in 19. chan_vpb was deprecated in 16, to be removed in 19. res_config_sqlite was deprecated in 16, to be removed in 19. res_monitor was deprecated in 16, to be removed in 21. conf2ael was deprecated in 16, to be removed in 19. muted was deprecated in 16, to be removed in 19. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29554 ASTERISK-29555 ASTERISK-29557 ASTERISK-29558 ASTERISK-29559 ASTERISK-29560 ASTERISK-29561 ASTERISK-29562 ASTERISK-29563 ASTERISK-29564 ASTERISK-29565 ASTERISK-29566 ASTERISK-29567 ASTERISK-29568 ASTERISK-29569 ASTERISK-29570 ASTERISK-29571 ASTERISK-29572 ASTERISK-29573 ASTERISK-29574 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Asterisk Development Team	4bf9b2d279	Update CHANGES and UPGRADE.txt for 16.20.0	4 years ago
Naveen Albert	af17973bda	func_frame_drop: New function Adds function to selectively drop specified frames in the TX or RX direction on a channel, including control frames. ASTERISK-29478 Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec	4 years ago
Alexander Traud	9a225692fe	aelparse: Accept an included context with timings. With Asterisk 1.6.0, in the main parser for the configuration file extensions.conf, the separator was changed from vertical bar to comma. However, the first separator was not changed in aelparse; it still had to be a vertical bar, and no comma was allowed. Additionally, this change allows the vertical bar for the first and last parameter again, even in the main parser, because the vertical bar was still accepted for the other parameters. ASTERISK-29540 Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81	4 years ago
Kevin Harwell	b2aab2891c	format_ogg_speex: Implement a "not supported" write handler This format did not specify a "write" handler, so when attempting to write to it (ast_writestream) a crash would occur. This patch adds a default handler that simply issues a "not supported" warning, thus no longer crashing. ASTERISK-29539 Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91	4 years ago
Naveen Albert	75c01f4bef	cdr_adaptive_odbc: Prevent filter warnings Previously, if CDR filters were used so that not all CDR records used all sections defined in cdr_adaptive_odbc.conf, then warnings will always be emitted (if each CDR record is unique to a particular section, n-1 warnings to be specific). This turns the offending warning log into a verbose message like the other one, since this behavior is intentional and not indicative of anything wrong. ASTERISK-29494 Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8	4 years ago
Naveen Albert	1298c02665	app_queue: Allow streaming multiple announcement files Allows multiple files comprising an agent announcement to be played by separating on the ampersand, similar to the multi-file support in other Asterisk applications. ASTERISK-29528 Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a	4 years ago
Igor Goncharovsky	8da1466478	res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request. It may be used to get all X- headers in case the actual set and names of headers unknown. ASTERISK-29389 Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b	4 years ago
Rijnhard Hessel	b40e97b1d7	res_statsd: handle non-standard meter type safely Meter types are not well supported, lacking support in telegraf, datadog and the official statsd servers. We deprecate meters and provide a compliant fallback for any existing usages. A flag has been introduced to allow meters to fallback to counters. ASTERISK-29513 Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7	4 years ago
Naveen Albert	0460e77e51	app_dtmfstore: New application to store digits Adds application to asynchronously collect digits dialed on a channel in the TX or RX direction using a framehook and stores them in a specified variable, up to a configurable number of digits. ASTERISK-29477 Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f	4 years ago
under	56e6486aa6	codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps. This makes the audio stream not-playable at the receiver side. Linphone isn't able to play such an audio - lots of disruptions are heard. Also I had complains of bad audio from users which use other types of phones. After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c). Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c). However, this flag is never set in Asterisk-12 and newer. Previously it has been set (see Asterisk-11). ASTERISK-29526 #close Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d	4 years ago
Joshua C. Colp	aec2087d25	docs: Remove embedded macro in WaitForCond XML documentation. Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c	4 years ago
Asterisk Development Team	b7f0d83698	Update CHANGES and UPGRADE.txt for 16.19.1	4 years ago
Kevin Harwell	7168aaf0dd	AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS If an SSL socket parent/listener was destroyed during the handshake, depending on timing, it was possible for the handling callback to attempt access of it after the fact thus causing a crash. ASTERISK-29415 #close Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d	4 years ago
Kevin Harwell	dbfde090ea	AST-2021-008 - chan_iax2: remote crash on unsupported media format If chan_iax2 received a packet with an unsupported media format, for example vp9, then it would set the frame's format to NULL. This could then result in a crash later when an attempt was made to access the format. This patch makes it so chan_iax2 now ignores/drops frames received with unsupported media format types. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	4 years ago
Joshua C. Colp	662e731b33	AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. If a re-INVITE is received after we have sent a BYE request then it is possible for no channel to be present on the session. If this occurs we allow PJSIP to produce the offer instead. Since the call is being hung up if it produces an incorrect offer it doesn't actually matter. This also ensures that code which produces SDP does not need to handle if a channel is not present. ASTERISK-29381 Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042	4 years ago
Andre Barbosa	e52fd18c80	res_stasis_playback: Check for chan hangup on play_on_channels Verify `ast_check_hangup` before looping to the next sound file. If the call is already hangup we just break the cycle. It also ensures that the PlaybackFinished event is sent if the call was hangup. This is also use-full when we are playing a big list of file for a channel that is hangup. Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played. With the patch we just break the playback cycle when the chan is hangup. ASTERISK-29501 #close Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8	4 years ago
Sean Bright	c86b4c573b	res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup. From RFC 8225 Section 5.2.1: The "dest" claim is a JSON object with the claim name of "dest" and MUST have at least one identity claim object. The "dest" claim value is an array containing one or more identity claim JSON objects representing the destination identities of any type (currently "tn" or "uri"). If the "dest" claim value array contains both "tn" and "uri" claim names, the JSON object should list the "tn" array first and the "uri" array second. Within the "tn" and "uri" arrays, the identity strings should be put in lexicographical order, including the scheme-specific portion of the URI characters. Additionally, make it clear that there was a failure to sign the JWT payload and not necessarily a memory allocation failure. Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9	4 years ago
Sebastien Duthil	acc3b8ba75	stun: Emit warning message when STUN request times out Without this message, it is not obvious that the reason is STUN timeout. ASTERISK-29507 #close Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74	4 years ago
Sean Bright	68dbcd5ef5	res_http_media_cache.c: Parse media URLs to find extensions. Use the URI parsing functions to parse playback URLs in order to find their file extensions. For backwards compatibility, we first look at the full URL, then at any Content-Type header, and finally at just the path portion of the URL. ASTERISK-27871 #close Change-Id: I16d0682f6d794be96539261b3e48f237909139cb	4 years ago
Sean Bright	f0fd0e7675	main/cdr.c: Correct Party A selection. This appears to just have been a copy/paste error from `6258bbe7`. Fix suggested by Ross Beer in ASTERISK~29166. Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37	4 years ago
Naveen Albert	cb67bc2ebb	app_reload: New Reload application Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	4 years ago
Sean Bright	6f135c0ef4	res_pjsip_config_wizard.c: Add port matching support. In `f8b0c2c9` we added support for port numbers in 'match' statements but neglected to include that support in the PJSIP config wizard. The removed code would have also prevented IPv6 addresses from being successfully used in the config wizard as well. ASTERISK-29503 #close Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d	4 years ago
Naveen Albert	7bc9a05059	app_waitforcond: New application While several applications exist to wait for a certain event to occur, none allow waiting for any generic expression to become true. This application allows for waiting for a condition to become true, with configurable timeout and checking interval. ASTERISK-29444 Change-Id: I08adf2824b8bc63405778cf355963b5005612f41	4 years ago
Andre Barbosa	884d863c08	res_stasis_playback: Send PlaybackFinish event only once for errors When we try to play a list of sound files in the same Play command, we get only one PlaybackFinish event, after all sounds are played. But in the case where the Play fails (because channel is destroyed for example), Asterisk will send one PlaybackFinish event for each sound file still to be played. If the list is big, Asterisk is sending many events. This patch adds a failed state so we can understand that the play failed. On that case we don't send the event, if we still have a list of sounds to be played. When we reach the last sound, we send the PlaybackFinish with the failed state. ASTERISK-29464 #close Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322	4 years ago
George Joseph	987eb8a5ea	jitterbuffer: Correct signed/unsigned mismatch causing assert If the system time has stepped backwards because of a time adjustment between the time a frame is timestamped and the time we check the timestamps in abstract_jb:hook_event_cb(), we get a negative interval, but we don't check for that there. abstract_jb:hook_event_cb() then calls fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed) and the first thing that does is assert(interval >= 0). There are several issues with this... * abstract_jb:hook_event_cb() saves the interval in a variable named "now" which is confusing in itself. * "now" is defined as an unsigned int which converts the negative value returned from ast_tvdiff_ms() to a large positive value. * fixed_jb_get()'s parameter is defined as a signed int so the interval gets converted back to a negative value. * fixed_jb_get()'s assert is NOT an ast_assert but a direct define that points to the system assert() so it triggers even in production mode. So... * hook_event_cb()'s "now" was renamed to "relative_frame_start" and changed to an int64_t. * hook_event_cb() now checks for a negative value right after retrieving both the current and framedata timestamps and just returns the frame if the difference is negative. * fixed_jb_get()'s local define of ASSERT() was changed to call ast_assert() instead of the system assert(). ASTERISK-29480 Reported by: Dan Cropp Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9	4 years ago
Naveen Albert	bee710121e	app_dial: Expanded A option to add caller announcement Hitherto, the A option has made it possible to play audio upon answer to the called party only. This option is expanded to allow for playback of an audio file to the caller instead of or in addition to the audio played to the answerer. ASTERISK-29442 Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e	4 years ago
Joshua C. Colp	4470838d89	core: Don't play silence for Busy() and Congestion() applications. When using the Busy() and Congestion() applications the function ast_safe_sleep is used by wait_for_hangup to safely wait on the channel. This function may send silence if Asterisk is configured to do so using the transmit_silence option. In a scenario where an answered channel dials a Local channel either directly or through call forwarding and the Busy() or Congestion() dialplan applications were executed with the transmit_silence option enabled the busy or congestion tone would not be heard. This is because inband generation of tones (such as busy and congestion) is stopped when other audio is sent to the channel they are being played to. In the given scenario the transmit_silence option would result in silence being sent to the channel, thus stopping the inband generation. This change adds a variant of ast_safe_sleep which can be used when silence should not be played to the channel. The wait_for_hangup function has been updated to use this resulting in the tones being generated as expected. ASTERISK-29485 Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133	4 years ago
Bernd Zobl	1b8f9a08d9	res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress With the fix for ASTERISK_28754 channels are no longer put on hold if an outbound INVITE is answered with a "Session Progress" containing "inactive" audio. The previous change moved the evaluation of the media attributes to `negotiate_incoming_sdp_stream()` to have the `remotely_held` status available when building the SDP in `create_outgoing_sdp_stream()`. This however means that an answer to an outbound INVITE, which does not traverse `negotiate_incoming_sdp_stream()`, cannot set the `remotely_held` status anymore. This change moves the check so that both, `negotiate_incoming_sdp_stream()` and `apply_negotiated_sdp_stream()` can do the checks. ASTERISK-29479 Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75	4 years ago

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