asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua C. Colp	93870e7bb4	policy: Deprecate modules and add versions to others. 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. app_macro was deprecated in 16, to be removed in 21. chan_sip was deprecated in 17, to be removed in 21. res_monitor was deprecated in 16, to be removed in 21. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29558 ASTERISK-29567 ASTERISK-29572 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Naveen Albert	6a89266b5b	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	8a6c9c3a76	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	049c7c1361	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	b5709e610e	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	0e023e6cf1	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	4f437ea1f4	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	728a52fb61	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	fa7d147e1b	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	de3f5350de	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
Sean Bright	6428124b06	res_http_media_cache: Cleanup audio format lookup in HTTP requests Asterisk first looks at the end of the URL to determine the file extension of the returned audio, which in many cases will not work because the URL may end with a query string or a URL fragment. If that fails, Asterisk then looks at the Content-Type header and then finally parses the URL to get the extension. The order has been changed such that we look at the Content-Type header first, followed by looking for the extension of the parsed URL. We no longer look at the end of the URL, which was error prone. ASTERISK-29527 #close Change-Id: I1e3f83b339ef2b80661704717c23568536511032	4 years ago
Joshua C. Colp	d0f189a5c9	docs: Remove embedded macro in WaitForCond XML documentation. Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c	4 years ago
Ben Ford	db7b025532	Update AMI and ARI versions for Asterisk 20. Bumped AMI and ARI versions for the next major Asterisk version (20). Change-Id: I2e65794f206d443178ab6895767fb53f04cc3e6a	4 years ago
Kevin Harwell	e8cda4b32c	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	1b62831f2c	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	ec16d2ecbd	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
Asterisk Development Team	e6ddbe0922	Update CHANGES and UPGRADE.txt for 19.0.0	4 years ago
Andre Barbosa	f4d3f021f9	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	d5bb27a06f	res_http_media_cache.c: Fix merge errors from 18 -> master ASTERISK-27871 #close Change-Id: I6624f2d3a57f76a89bb372ef54a124929a0338d7	4 years ago
Sean Bright	237285a9a8	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
Sean Bright	d568326807	res_http_media_cache.c: Parse media URLs to find extensions. Use cURL's URL parsing API, falling back to the urlparser library, 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	785e4afc20	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
Sebastien Duthil	8a21d466ea	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
Naveen Albert	244491f9b2	app_reload: New Reload application Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	4 years ago
Igor Goncharovsky	99d44f0c5a	res_ari: Fix audiosocket segfault Add check that data parameter specified when audiosocket used for externalMedia. ASTERISK-29514 #close Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617	4 years ago
Sean Bright	0ac9c83561	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	c01b4e0d4b	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	a47308ccb2	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	bc973bd719	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	1e5a2cfe30	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	5382b9dbb8	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	c30f68a57b	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
George Joseph	b7027de195	res_pjsip_messaging: Overwrite user in existing contact URI When the MessageSend destination is in the form PJSIP/<number>@<endpoint> and the endpoint's contact URI already has a user component, that user component will now be replaced with <number> when creating the request URI. ASTERISK_29404 Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5	4 years ago
Bernd Zobl	f160725fc4	res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter Set preferred transport when querying the local address to use in filter_on_tx_messages(). This prevents the module to erroneously select the wrong transport if more than one transports of the same type (TCP or TLS) are configured. ASTERISK-29241 Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6	4 years ago
Naveen Albert	f812c57477	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	4 years ago
Jaco Kroon	56c2cc474b	func_lock: Add "dialplan locks show" cli command. For example: arthur*CLI> dialplan locks show func_lock locks: Name Requesters Owner uls-autoref 0 (unlocked) 1 total locks listed. Obviously other potentially useful stats could be added (eg, how many times there was contention, how many times it failed etc ... but that would require keeping the stats and I'm not convinced that's worth the effort. This was useful to troubleshoot some other issues so submitting it. Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	19a8383a1f	func_lock: Prevent module unloading in-use module. The scenario where a channel still has an associated datastore we cannot unload since there is a function pointer to the destroy and fixup functions in play. Thus increase the module ref count whenever we allocate a datastore, and decrease it during destroy. In order to tighten the race that still exists in spite of this (below) add some extra failure cases to prevent allocations in these cases. Race: If module ref is zero, an LOCK or TRYLOCK is invoked (near) simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if in such a case the datastore is created prior to unloading being set to true (first step in module unload) then it's possible that the module will unload with the destructor being called (and segfault) post the module being unloaded. The module will however wait for such locks to release prior to unloading. If post that we can recheck the module ref before returning the we can (in theory, I think) eliminate the last of the race. This race is mostly theoretical in nature. Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	e8875d5ca1	func_lock: Fix memory corruption during unload. AST_TRAVERSE accessess current as current = current->(field).next ... and since we free current (and ast_free poisons the memory) we either end up on a ast_mutex_lock to a non-existing lock that can never be obtained, or a segfault. Incidentally add logging in the "we have to wait for a lock to release" case, and remove an ineffective statement that sets memory that was just cleared by ast_calloc to zero. Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	caceba7988	func_lock: Fix requesters counter in error paths. In two places we bail out with failure after we've already incremented the requesters counter, if this occured then it would effectively result in unload to wait indefinitely, thus preventing clean shutdown. Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Naveen Albert	b742514553	app_originate: Allow setting Caller ID and variables Caller ID can now be set on the called channel and Variables can now be set on the destination using the Originate application, just as they can be currently using call files or the Manager Action. ASTERISK-29450 Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66	4 years ago
Sean Bright	c0fc8adbb6	menuselect: Fix description of several modules. The text description needs to be the last thing on the AST_MODULE_INFO line to be pulled in properly by menuselect. Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832	4 years ago
Naveen Albert	35437879e5	app_confbridge: New ConfKick() application Adds a new ConfKick() application, which may be used to kick a specific channel, all channels, or all non-admin channels from a specified conference bridge, similar to existing CLI and AMI commands. ASTERISK-29446 Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b	4 years ago
Naveen Albert	1b38e89734	res_pjsip_dtmf_info: Hook flash Adds hook flash recognition support for application/hook-flash. ASTERISK-29460 Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea	4 years ago
Naveen Albert	5f8cabc232	app_confbridge: New option to prevent answer supervision A new user option, answer_channel, adds the capability to prevent answering the channel if it hasn't already been answered yet. ASTERISK-29440 Change-Id: I26642729d0345f178c7b8045506605c8402de54b	4 years ago
Naveen Albert	c8bf8a54c2	sip_to_pjsip: Fix missing cases Adds the "auto" case which is valid with both chan_sip dtmfmode and chan_pjsip's dtmf_mode, adds subscribecontext to subscribe_context conversion, and accounts for cipher = ALL being invalid. ASTERISK-29459 Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2	4 years ago
George Joseph	c3654a9959	res_pjsip_messaging: Refactor outgoing URI processing * Implemented the new "to" parameter of the MessageSend() dialplan application. This allows a user to specify a complete SIP "To" header separate from the Request URI. * Completely refactored the get_outbound_endpoint() function to actually handle all the destination combinations that we advertized as supporting. * We now also accept a destination in the same format as Dial()... PJSIP/number@endpoint * Added lots of debugging. ASTERISK-29404 Reported by Brian J. Murrell Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce	4 years ago
Naveen Albert	eeffad1b62	func_math: Three new dialplan functions Introduces three new dialplan functions, MIN and MAX, which can be used to calculate the minimum or maximum of up to two numbers, and ABS, an absolute value function. ASTERISK-29431 Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d	4 years ago
Ben Ford	12e8600849	STIR/SHAKEN: Add Date header, dest->tn, and URL checking. STIR/SHAKEN requires a Date header alongside the Identity header, so that has been added. Still on the outgoing side, we were missing the dest->tn section of the JSON payload, so that has been added as well. Moving to the incoming side, URL checking has been added to the public cert URL to ensure that it starts with http. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c	4 years ago
Joshua C. Colp	44fde9f428	res_pjsip: On partial transport reload also move factories. For connection oriented transports PJSIP uses factories to produce transports. When doing a partial transport reload we need to also move the factory of the transport over so that anything referencing the transport (such as an endpoint) has the factory available. ASTERISK-29441 Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161	4 years ago
Naveen Albert	19b5097d87	func_volume: Add read capability to function. Up until now, the VOLUME function has been write only, so that TX/RX values can be set but not read afterwards. Now, previously set TX/RX values can be read later. ASTERISK-29439 Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f	4 years ago

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