asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	178cb0ffe4	res: Fix for Doxygen. These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d	4 years ago
Dustin Marquess	b2e71b82e7	res_fax_spandsp: Add spandsp 3.0.0+ compatibility Newer versions of spandsp did refactoring of code to add new features like color FAXing. This refactoring broke backwards compatibility. Add support for the new version while retaining support for 0.0.6. ASTERISK-29729 #close Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1	4 years ago
Alexander Traud	20d9158c9c	main: Fix for Doxygen. ASTERISK-29763 Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e	4 years ago
Alexander Traud	f946b92553	progdocs: Fix for Doxygen, the hidden parts. ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b	4 years ago
Naveen Albert	bcb7aee723	documentation: Standardize examples Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b	4 years ago
Alexander Traud	b290bb1251	stir/shaken: Avoid a compiler extension of GCC. ASTERISK-29776 Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73	4 years ago
Naveen Albert	ca2e13e18f	res_tonedetect: Add call progress tone detection Makes basic call progress tone detection available in a tech-agnostic manner with the addition of the ToneScan application. This can determine if the channel has encountered a busy signal, SIT tones, dial tone, modem, fax machine, etc. A few basic async progress tone detect options are also added to the TONE_DETECT function. ASTERISK-29720 #close Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90	4 years ago
Alexander Traud	783b775946	odbc: Fix for Doxygen. ASTERISK-29754 Change-Id: Ia09eb68d283d201d9a6fbeccfc0efe83fe0502a5	4 years ago
Alexander Traud	c549eda0a7	parking: Fix for Doxygen. ASTERISK-29753 Change-Id: I7a61974584f6169502e6860fc711919fe7bbfaa7	4 years ago
Alexander Traud	5b5a9ea4f0	res_ari: Fix for Doxygen. ASTERISK-29756 Change-Id: I2f1c1eea1c902492b77b74de9950f20ebbb7e758	4 years ago
Alexander Traud	6988386234	stasis: Fix for Doxygen. ASTERISK-29750 Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9	4 years ago
Alexander Traud	31c26fcbc6	res_xmpp: Fix for Doxygen. ASTERISK-29749 Change-Id: I7885793b63bdeaa883e76edb899bbba9660eb1c5	4 years ago
Alexander Traud	bae495601a	res_pjsip: Fix for Doxygen. ASTERISK-29747 Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084	4 years ago
Alexander Traud	44a9c16e9c	progdocs: Avoid 'name' with Doxygen \file. Fixes four misuses 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: I0c18b4c863c6988b138c77448057349a9ee7052d	4 years ago
Naveen Albert	1cd2584b27	res_pjsip_callerid: Fix OLI parsing Fix parsing of ANI2/OLI information, since it was previously parsing the user, when it should have been parsing other_param. Also improves the parsing by using pjproject native functions rather than trying to parse the parameters ourselves like chan_sip did. A previous attempt at this caused a crash, but this works correctly now. ASTERISK-29703 #close Change-Id: I8f3c79032d9ea1a21d16f8e11f22bd8d887738a1	4 years ago
Josh Soref	dcf492e7b6	res: Spelling fixes Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Ben Ford	2e55c0fded	STIR/SHAKEN: Option split and response codes. The stir_shaken configuration option now has 4 different choices to pick from: off, attest, verify, and on. Off and on behave the same way they do now. Attest will only perform attestation on the endpoint, and verify will only perform verification on the endpoint. Certain responses are required to be sent based on certain conditions for STIR/SHAKEN. For example, if we get a Date header that is outside of the time range that is considered valid, a 403 Stale Date response should be sent. This and several other responses have been added. Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7	4 years ago
Kevin Harwell	859f579504	res_speech: Add a type conversion, and new engine unregister methods Add a new function that converts a speech results type to a string. Also add another function to unregister an engine, but returns a pointer to the unregistered engine object instead of a success/fail integer. Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192	4 years ago
Matthew Kern	15e432220c	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
Jean Aunis	0ab4e7491d	res_rtp_asterisk: fix memory leak Add missing reference decrement in rtp_deallocate_transport() ASTERISK-29671 Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9	4 years ago
Joseph Nadiv	4368764032	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	ea36473c45	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
Sean Bright	b2c834e349	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
Guido Falsi	03377c35fc	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
Naveen Albert	1a23c9c047	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
Sungtae Kim	d9747104ff	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	a6eb1b6f95	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	2806a45034	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
Jasper Hafkenscheid	c1a575907b	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
sungtae kim	479cc17f45	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
Sebastien Duthil	ac492f2ff8	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
Alexander Traud	82d6bd7ec9	res_rtp_asterisk: sqrt(.) requires the header math.h. ASTERISK-29616 Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900	4 years ago
Alexander Traud	8410afc7ab	dialplan: Add one static and fix two whitespace errors. Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83	4 years ago
George Joseph	b72425b1f0	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
Alexander Traud	a8e8b3aaff	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	121860e3f6	mgcp: Remove dead debug code ASTERISK-20339 #close Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5	4 years ago
Joshua C. Colp	13fd0789a2	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
Igor Goncharovsky	1e4ed61a2b	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	71dd1d91ad	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
Joshua C. Colp	523a795289	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	2c3defc6c6	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	30feaadabf	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	76c09b1cfd	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
Igor Goncharovsky	b9bb96ffed	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	146b59df3f	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
Andre Barbosa	283812e492	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
Bernd Zobl	6b041d1092	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	702e1d33b5	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	804788037e	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
Sean Bright	26059f8616	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

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5250 Commits (6c9e8afd4ea55a33680ee580658230b51f307292)