6614 Commits (5221b321fdf367aae7ecaaa9934f06e131f8df63)

Author	SHA1	Message	Date
Alexander Traud	ed851cbba8	channel: Fix for Doxygen. ... ASTERISK-29751 Change-Id: Ie04da5029c57ebee44733bdf05013156abe80176	5 years ago
Alexander Traud	6b0cee202b	stasis: Fix for Doxygen. ... ASTERISK-29750 Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9	5 years ago
Alexander Traud	773a60cda6	app: Fix for Doxygen. ... ASTERISK-29752 Change-Id: If40cbd01d47a6cfd620b18206dedb8460216c8af	5 years ago
Alexander Traud	94d8d124fe	bridge_channel: Fix for Doxygen. ... ASTERISK-29736 Change-Id: Ia5370289e6526001a6b52754b533bcea1a9d7e5c	5 years ago
Alexander Traud	79cb1cdb65	progdocs: Avoid 'name' with Doxygen \file. ... Fixes one misuse of the parameter 'name'. Additionally, for consistency and to avoid such an issue in future, those few other places, which used '\file name', were changed just to '\file'. Then, Doxygen uses the name of the current file. ASTERISK-29733 Change-Id: I1a915713d39223b03532549e6c16a98cefb4fc23	5 years ago
Alexander Traud	94575f9a9a	progdocs: Use Doxygen \example correctly. ... ASTERISK-29734 Change-Id: I83b51e85cd71867645ab3a8a820f8fd1f065abd2	5 years ago
Naveen Albert	ac2c984f22	app_read: Fix custom terminator functionality regression ... Currently, when the t option is specified with no arguments, the # character is still treated as a terminator, even though no character should be treated as a terminator. This is because a previous regression fix was modified to remove the use of NULL as a default altogether. However, NULL and an empty string actually refer to different arrangements and should be treated differently. NULL is the default terminator (#), while an empty string removes the terminator altogether. This is the behavior being used by the rest of the core. Additionally, since S_OR catches empty strings as well as NULL (not intended), this is changed to a ternary operator instead, which fixes the behavior. ASTERISK-29705 #close Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086	5 years ago
Josh Soref	47c50aa47e	main: Spelling fixes ... Correct typos of the following word families: analysis nuisance converting although transaction desctitle acquire update evaluate thousand this dissolved management integrity reconstructed decrement further on irrelevant currently constancy anyway unconstrained featuregroups right larger evaluated encumbered languages digits authoritative framing blindxfer tolerate traverser exclamation perform permissions rearrangement performing processing declension happily duplicate compound hundred returns elicit allocate actually paths inheritance atxferdropcall earlier synchronization multiplier acknowledge across against thousands joyous manipulators guaranteed emulating soundfile ASTERISK-29714 Change-Id: I926ba4b11e9f6dd3fdd93170ab1f9b997910be70	5 years ago
Sean Bright	74ad80609e	pbx.c: Don't remove dashes from hints on reload. ... When reloading dialplan, hints created dynamically would lose any dash characters. Now we ignore those dashes if we are dealing with a hint during a reload. ASTERISK-28040 #close Change-Id: I95e48f5a268efa3c6840ab69798525d3dce91636	5 years ago
Alexander Traud	45266c66d5	stasis: Avoid 'dispatched' as unused variable in normal mode. ... ASTERISK-29710 Change-Id: Ia849f1172e4e694c5d5d7f0cad449f936ee12216	5 years ago
Sean Bright	11f291f6f0	various: Fix GCC 11.2 compilation issues. ... * Initialize some variables that are never used anyway. * Use valid pointers instead of integers cast to void pointers when calling pthread_setspecific(). ASTERISK-29711 #close ASTERISK-29713 #close Change-Id: I8728cd6f2f4b28e0e48113c5da450b768c2a6683	5 years ago
Kevin Harwell	d112916e98	strings/json: Add string delimter match, and object create with vars methods ... Add a function to check if there is an exact match a one string between delimiters in another string. Add a function that will create an ast_json object out of a list of Asterisk variables. An excludes string can also optionally be passed in. Also, add a macro to make it easier to get object integers. Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226	5 years ago
Sebastien Duthil	fe0b5aed7b	main/stun.c: fix crash upon STUN request timeout ... Some ast_stun_request users do not provide a destination address when sending to a connection-mode socket. ASTERISK-29691 Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc	5 years ago
Shloime Rosenblum	e8ad58217f	main/say.c: Support future dates with Q and q format params ... The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today ASTERISK-29637 Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02	5 years ago
Sean Bright	97ce647afd	message.c: Support 'To' header override with AMI's MessageSend. ... The MessageSend AMI action has been updated to allow the Destination and the To addresses to be provided separately. This brings the MessageSend manager command in line with the capabilities of the MessageSend dialplan application. ASTERISK-29663 #close Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c	5 years ago
Naveen Albert	41ba9f5f31	logger: Add custom logging capabilities ... Adds the ability for users to log to custom log levels by providing custom log level names in logger.conf. Also adds a logger show levels CLI command. ASTERISK-29529 Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702	5 years ago
Naveen Albert	11516e4b8e	func_sayfiles: Retrieve say file names ... Up until now, all of the logic used to translate arguments to the Say applications has been directly coupled to playback, preventing other modules from using this logic. This refactors code in say.c and adds a SAYFILES function that can be used to retrieve the file names that would be played. These can then be used in other applications or for other purposes. Additionally, a SayMoney application and a SayOrdinal application are added. Both SayOrdinal and SayNumber are also expanded to support integers greater than one billion. ASTERISK-29531 Change-Id: If9718c89353b8e153d84add3cc4637b79585db19	5 years ago
Naveen Albert	698604a064	res_tonedetect: Tone detection module ... dsp.c contains arbitrary tone detection functionality which is currently only used for fax tone recognition. This change makes this functionality publicly accessible so that other modules can take advantage of this. Additionally, a WaitForTone and TONE_DETECT app and function are included to allow users to do their own tone detection operations in the dialplan. ASTERISK-29546 Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26	5 years ago
Sean Bright	7a7b9e47ca	term.c: Add support for extended number format terminfo files. ... ncurses 6.1 introduced an extended number format for terminfo files which the terminfo parsing in Asterisk is not able to parse. This results in some TERM values that do support color (screen-256color on Ubuntu 20.04 for example) to not get a color console. ASTERISK-29630 #close Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3	5 years ago
Sean Bright	fb72158f46	dns.c: Load IPv6 DNS resolvers if configured. ... IPv6 nameserver addresses are stored in different part of the __res_state structure, so look there if we appear to have support for it. ASTERISK-28004 #close Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 years ago
Naveen Albert	80be0edae7	pbx_builtins: Corrects SayNumber warning ... Previously, SayNumber always emitted a warning if the caller hung up during execution. Usually this isn't correct, so check if the channel hung up and, if so, don't emit a warning. ASTERISK-29475 Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594	5 years ago
Joshua C. Colp	0ee56cc2bd	asterisk: We've moved to Libera Chat! ... Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575	5 years ago
Naveen Albert	ea117be4c6	AMI: Add AMI event to expose hook flash events ... Although Asterisk can receive and propogate flash events, it currently provides no mechanism for doing anything with them itself. This AMI event allows flash events to be processed by Asterisk. Additionally, AST_CONTROL_FLASH is included in a switch statement in channel.c to avoid throwing a warning when we shouldn't. ASTERISK-29380 Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81	5 years ago
Naveen Albert	859cd2a56b	main/file.c: Don't throw error on flash event. ... AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c where it should be ignored. Adding this to the switch ensures a warning isn't thrown on RFC2833 flash events, since nothing's amiss. ASTERISK-29372 Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc	5 years ago
Ben Ford	0b4b207076	STIR/SHAKEN: Switch to base64 URL encoding. ... STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	5 years ago
George Joseph	63c25d3821	Updates for the MessageSend Dialplan App ... Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	5 years ago
Sean Bright	d72c7d2d66	translate.c: Avoid refleak when checking for a translation path ... Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6	5 years ago
Joshua C. Colp	d2b8141d26	chan_local: Skip filtering audio formats on removed streams. ... When a stream topology is provided to chan_local when dialing it filters the audio formats down. This operation did not skip streams which were removed (that have no formats) resulting in calling being aborted. This change causes such streams to be skipped. ASTERISK-29407 Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6	5 years ago
Jean Aunis	080d0eb72c	translate.c: Take sampling rate into account when checking codec's buffer size ... Up/down sampling changes the number of samples produced by a translation. This must be taken into account when checking the codec's buffer size. ASTERISK-29328 Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055	5 years ago
George Joseph	a4a63db253	bridge_channel_write_frame: Check for NULL channel ... There is a possibility, when bridge_channel_write_frame() is called, that the bridge_channel->chan will be NULL. The first thing bridge_channel_write_frame() does though is call ast_channel_is_multistream() which had no check for a NULL channel and therefore caused a segfault. Since it's still possible for bridge_channel_write_frame() to write the frame to the other channels in the bridge, we don't want to bail before we call ast_channel_is_multistream() but we can just skip the multi-channel stuff. So... bridge_channel_write_frame() only calls ast_channel_is_multistream() if bridge_channel->chan is not NULL. As a safety measure, ast_channel_is_multistream() now returns false if the supplied channel is NULL. ASTERISK-29379 Reported-by: Vyrva Igor Reported-by: Ross Beer Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce	5 years ago
Sean Bright	84115fe657	loader.c: Speed up deprecation metadata lookup ... Only use an XPath query once per module, then just navigate the DOM for everything else. Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92	5 years ago
Joshua C. Colp	a9e1e0e1c8	loader: Output warnings for deprecated modules. ... Using the information from the MODULEINFO XML we can now output useful information at the end of module loading for deprecated modules. This includes the version it was deprecated in, the version it will be removed in, and the replacement if available. ASTERISK-29339 Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a	5 years ago
Kevin Harwell	20af6d23df	time: Add timeval create and unit conversion functions ... Added a TIME_UNIT enumeration, and a function that converts a string to one of the enumerated values. Also, added functions that create and initialize a timeval object using a specified value, and unit type. Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392	5 years ago
Mark Murawski	98d149b4ce	logger: Console sessions will now respect logger.conf dateformat= option ... The 'core' console (ie: asterisk -c) does read logger.conf and does use the dateformat= option. Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf and uses a hard coded dateformat option for printing received verbose messages: main/logger.c: static char dateformat[256] = "%b %e %T" This change will load logger.conf for each remote console session and use the dateformat= option to set the per-line timestamp for verbose messages Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1 ASTERISK-25358: #close Reported-by: Igor Liferenko	5 years ago
Joshua C. Colp	73a52c53ce	core_unreal: Fix deadlock with T.38 control frames. ... When using the ast_unreal_lock_all function no channel locks can be held before calling it. This change unlocks the channel that indicate was called on before doing so and then relocks it afterwards. ASTERISK-29035 Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9	5 years ago
Joshua C. Colp	ceb8404667	channel: Fix crash in suppress API. ... There exists an inconsistency with framehook usage such that it is only on reads that the frame should be freed, not on writes as well. ASTERISK-29071 Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472	6 years ago
Joshua C. Colp	11b53aecc8	sorcery: Add support for more intelligent reloading. ... Some sorcery objects actually contain dynamic content that can change despite the underlying configuration itself not changing. A good example of this is the res_pjsip_endpoint_identifier_ip module which allows specifying hostnames. While the configuration may not change between reloads the DNS information of the hostnames can. This change adds the ability for a sorcery object to be marked as having dynamic contents which is then taken into account when reloading by the sorcery file based config module. If there is an object with dynamic content then a reload will be forced while if there are none then the existing behavior of not reloading occurs. ASTERISK-29321 Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1	6 years ago
Joshua C. Colp	4083aa6546	asterisk: Update copyright. ... ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	6 years ago