34066 Commits (d6ce7bb395cdc0f15a3e537d9a3daf95f382c40c)

 

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Author	SHA1	Message	Date
Sean Bright	44fd75fae2	configure: Remove unused OpenSSL SRTP check. ... Discovered while looking at ASTERISK~29684. Usage was removed in change I3c77c7b00b2ffa2e935632097fa057b9fdf480c0. Change-Id: Iaf2f7a16ea5a7eee6375319347e4b40b8e7b10e3	4 years ago
Mike Bradeen	072f2ebb12	build: prevent binary downloads for non x86 architectures ... download_externals: Add check for i686 and i386 (in addition to the current x86_64) and exit if not one of the three. ASTERISK-26497 Change-Id: Ia4d429fcefa5b2f5b6e99159d4607de8e8325b2f	4 years ago
Sebastien Duthil	51859252f7	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	4 years ago
Sean Bright	9fcd50a8c9	Makefile: Use basename in a POSIX-compliant way. ... If you aren't using GNU coreutils, chances are that your basename doesn't know about the -s argument. Luckily for us, basename does what we need it do even without the -s argument. Change-Id: I8b81a429bb037b997ee6640ff8a2b5e860962bb7	4 years ago
Mark Murawski	7fc26e8617	pbx_ael: Fix crash and lockup issue regarding 'ael reload' ... Avoid infinite recursion and crash Change-Id: I8ed05ec3aa2806c50c77edc5dd0cd4e4fa08b3f4	4 years ago
Naveen Albert	7ff6c43760	chan_iax2: Add encryption for RSA authentication ... Adds support for encryption to RSA-authenticated calls. Also prevents crashes if an RSA IAX2 call is initiated to a switch requiring encryption but no secret is provided. ASTERISK-20219 Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40	4 years ago
Matthew Kern	5e9799a42e	res_pjsip_t38: bind UDPTL sessions like RTP ... In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the fallback use of the transport's bind address solve problems sending media on systems that cannot send ipv4 packets on ipv6 sockets, and certain other situations. This change extends both of these behaviors to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific problems on these systems, introducing a new option endpoint/t38_bind_udptl_to_media_address. ASTERISK-29402 Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557	4 years ago
Naveen Albert	b40ca38c56	app_read: Fix null pointer crash ... If the terminator character is not explicitly specified and an indications tone is used for reading a digit, there is no null pointer check so Asterisk crashes. This prevents null usage from occuring. ASTERISK-29673 #close Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac	4 years ago
Jean Aunis	6bc747b639	res_rtp_asterisk: fix memory leak ... Add missing reference decrement in rtp_deallocate_transport() ASTERISK-29671 Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9	4 years ago
Shloime Rosenblum	d20587250e	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	4 years ago
Joseph Nadiv	47cb177baf	res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts ... The behavior of max_contacts and remove_existing are connected. If remove_existing is enabled, the soonest expiring contacts are removed. This may occur when there is an unavailable contact. Similarly, when remove_existing is not enabled, registrations from good endpoints are rejected in favor of retaining unavailable contacts. This commit adds a new AOR option remove_unavailable, and the effect of this setting will depend on remove_existing. If remove_existing is set to no, we will still remove unavailable contacts when they exceed max_contacts, if there are any. If remove_existing is set to yes, we will prioritize the removal of unavailable contacts before those that are expiring soonest. ASTERISK-29525 Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784	4 years ago
Joshua C. Colp	0aac38c0ac	ari: Ignore invisible bridges when listing bridges. ... When listing bridges we go through the ones present in ARI, get their snapshot, turn it into JSON, and add it to the payload we ultimately return. An invisible "dial bridge" exists within ARI that would also try to be added to this payload if the channel "create" and "dial" routes were used. This would ultimately fail due to invisible bridges having no snapshot resulting in the listing of bridges failing. This change makes it so that the listing of bridges ignores invisible ones. ASTERISK-29668 Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a	4 years ago
Naveen Albert	d900130021	func_vmcount: Add support for multiple mailboxes ... Allows multiple mailboxes to be specified for VMCOUNT instead of just one. ASTERISK-29661 #close Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813	4 years ago
Sean Bright	5ca9898dfb	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	4 years ago
Naveen Albert	de6ecd5e34	func_channel: Add CHANNEL_EXISTS function. ... Adds a function to check for the existence of a channel by name or by UNIQUEID. ASTERISK-29656 #close Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb	4 years ago
Naveen Albert	5abf499d23	app_queue: Fix hint updates for included contexts ... Previously, if custom hints were used with the hint: format in app_queue, when device state changes occured, app_queue would only do a literal string comparison of the context used for the hint in app_queue and the context of the hint which just changed state. This caused hints to not update and become stale if the context associated with the agent included the context which actually changes state, essentially completely breaking device state for any such agents defined in this manner. This fix adds an additional check to ensure that included contexts are also compared against the context which changed state, so that the behavior is correct no matter whether the context is specified to app_queue directly or indirectly. ASTERISK-29578 #close Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78	4 years ago
Sean Bright	02f54e2751	res_http_media_cache.c: Compare unaltered MIME types. ... Rather than stripping parameters from Content-Type headers before comparison, first try to compare the whole string. If no match is found, strip the parameters and try that way. ASTERISK-29275 #close Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f	4 years ago
Naveen Albert	148f8355a0	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	4 years ago
Sean Bright	6698753b24	app_externalivr.c: Fix mixed leading whitespace in source code. ... No functional changes. Change-Id: I46514152c0af67f395526374aaa847ccd6a85378	4 years ago
Guido Falsi	29ad5b18f1	res_rtp_asterisk.c: Fix build failure when not building with pjproject. ... Some code has been added referencing symbols defined in a block protected by #ifdef HAVE_PJPROJECT. Protect those code parts in ifdef blocks too. ASTERISK-29660 Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f	4 years ago
George Joseph	54a9dbb2b8	pjproject: Add patch to fix trailing whitespace issue in rtpmap ... An issue was found where a particular manufacturer's phones add a trailing space to the end of the rtpmap attribute when specifying a payload type that has a "param" after the format name and clock rate. For example: a=rtpmap:120 opus/48000/2 \r\n Because pjmedia_sdp_attr_get_rtpmap currently takes everything after the second '/' up to the line end as the param, the space is included in future comparisons, which then fail if the param being compared to doesn't also have the space. We now use pj_scan_get() to parse the param part of rtpmap so trailing whitespace is automatically stripped. ASTERISK-29654 Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b	4 years ago
Carlos Oliva	07c297d058	app_mp3: Force output to 16 bits in mpg123 ... In new mpg123 versions (since 1.26) the default output is 32 bits Asterisk expects the output in 16 bits, so we force the output to be on 16 bits. It will work wit new and old versions of mpg123. Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key! ASTERISK-29635 #close Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816	4 years ago
Naveen Albert	5b5c358e4b	res_pjsip_caller_id: Add ANI2/OLI parsing ... Adds parsing of ANI II digits (Originating Line Information) to PJSIP, on par with what currently exists in chan_sip. ASTERISK-29472 Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847	4 years ago
Naveen Albert	b760bad2b9	app_mf: Add channel agnostic MF sender ... Adds a SendMF application and PlayMF manager event to send arbitrary R1 MF tones on the current or specified channel. ASTERISK-29496 Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4	4 years ago
Naveen Albert	18c92353f8	app_stack: Include current location if branch fails ... Previously, the error emitted when app_stack tries to branch to a dialplan location that doesn't exist has included only the information about the attempted branch in the error log. This adds the current location as well so users can see where the branch failed in the logs. ASTERISK-29626 Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce	4 years ago
Sean Bright	46afd61b75	test_http_media_cache.c: Fix copy/paste error during test deregistration. ... Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c	4 years ago
Sungtae Kim	a1fa8df0ae	resource_channels.c: Fix external media data option ... Fixed the external media creation handle to handle the 'data' option correctly. ASTERISK-29629 Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129	4 years ago
Naveen Albert	b8fc77a35b	func_strings: Add STRBETWEEN function ... Adds the STRBETWEEN function, which can be used to insert a substring between each character in a string. For instance, this can be used to insert pauses between DTMF tones in a string of digits. ASTERISK-29627 Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258	4 years ago
Sean Bright	c4037d4aa3	test_abstract_jb.c: Fix put and put_out_of_order memory leaks. ... We can't rely on RAII_VAR(...) to properly clean up data that is allocated within a loop. ASTERISK-27176 #close Change-Id: Ib575616101230c4f603519114ec62ebf3936882c	4 years ago
Naveen Albert	e0111a56fa	func_env: Add DIRNAME and BASENAME functions ... Adds the DIRNAME and BASENAME functions, which are wrappers around the corresponding C library functions. These can be used to safely and conveniently work with file paths and names in the dialplan. ASTERISK-29628 #close Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3	4 years ago
Naveen Albert	ddf6299b8d	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	4 years ago
Naveen Albert	7df69633cf	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	4 years ago
George Joseph	448962d056	res_snmp: Add -fPIC to _ASTCFLAGS ... With gcc 11, res/res_snmp.c and res/snmp/agent.c need the -fPIC option added to its _ASTCFLAGS. ASTERISK-29634 Change-Id: I34649c85e075fd954e578378fabf798c3f038f50	4 years ago
Sean Bright	26fc5f3c72	app_voicemail.c: Ability to silence instructions if greeting is present. ... There is an option to silence voicemail instructions but it does not take into consideration if a recorded greeting exists or not. Add a new 'S' option that does that. ASTERISK-29632 #close Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4	4 years ago
Sean Bright	605dd03b36	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	4 years ago
Jasper Hafkenscheid	c07d531191	res_srtp: Disable parsing of not enabled cryptos ... When compiled without extended srtp crypto suites also disable parsing these from received SDP. This prevents using these, as some client implementations are not stable. ASTERISK-29625 Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a	4 years ago
Sean Bright	695fc3dbd7	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	4 years ago
George Joseph	976521c9a2	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	79d6d222d6	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	5029e78f39	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	3072c540bb	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	bbf4f30059	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	6cc004dc5a	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	6fbf55ac11	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	f01a0398f8	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
Naveen Albert	92f9ae32a8	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
Alexander Traud	63d27af3ca	res_rtp_asterisk: sqrt(.) requires the header math.h. ... ASTERISK-29616 Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900	4 years ago
Alexander Traud	fbdd8a7f8a	dialplan: Add one static and fix two whitespace errors. ... Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83	4 years ago
Sarah Autumn	466eb4a52b	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	c4839c04b6	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