asterisk

Commit Graph

Author	SHA1	Message	Date
Nicholas John Koch	fef97a9a72	res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings A warning was triggered that there may be a problem regarding file extension (which is correct and should not be set anyway). The warning also appeared if there was dot within the path itself. E.g. [sales-queue-hold] mode=playlist entry=/var/www/domain.tld/moh/funky_music The music played correctly but you get a warning message. Now there will be a check if the position of a potential dot character is after the last position of a slash character. This dot charachter will be treated as a extension naming. Dots within the path then ignored. ASTERISK-28892 Reported-By: Nicholas John Koch Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e	6 years ago
sungtae kim	c8c94b6cf1	res_rtp_asterisk.c: Fixed memory leak Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	6 years ago
Joshua C. Colp	15cbff9d54	ari: Allow variables to be set on channel create. This change adds the same variable functionality that is available for originating a channel to the create call. Now when creating a channel you can specify dialplan variables to set instead of having to do another API call. ASTERISK-28896 Change-Id: If13997ba818136d7c070585504fc4164378aa992	6 years ago
Roger James	c8dec423d2	pjsip_resolver.c: Ensure AAAA dns requests are made. 1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups when an IPV6 transport type has been requested. 2. Rename all occurrences of pjsip_transport_get_type_name to pjsip_transport_get_type_desc. This ensures that the log/debug info shows whether the transport is IPv6 or IPv4. 3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport types. This results in invalid values. Use a bitwise or instead. ASTERISK-26780 Patches: pjsip_resolver.c uploaded by Peter Sokolov (License #7070) Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58	6 years ago
Ben Ford	e29df34de0	res_stir_shaken: Added dialplan function and API call. Adds the "STIR_SHAKEN" dialplan function and an API call to add a STIR_SHAKEN verification result to a channel. This information will be held in a datastore on the channel that can later be queried through the "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results including identity, attestation, and verify_result. Here are some examples: STIR_SHAKEN(count) STIR_SHAKEN(0, identity) STIR_SHAKEN(1, attestation) STIR_SHAKEN(2, verify_result) Getting the count can be used to iterate through the results and pull information by specifying the index and the field you want to retrieve. Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82	6 years ago
Guido Falsi	801d570f6e	pjproject: Fix race condition when building with parallel make Pjproject makefiles miss some dependencies which can cause race conditions when building with parallel make processes. This patch adds such dependencies correctly. ASTERISK-28879 #close Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru> Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22	6 years ago
Roger James	4a072c4890	res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered. Changed source and destination address fields in struct pjsip_history_entry so that they are long enough to hold an IPv6 address. ASTERISK-28854 Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e	6 years ago
traud	f9ea75d117	tcptls: Fix notice when TLS is enabled but not supported. ASTERISK-28797 Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28	6 years ago
traud	527e4f6542	app_osplookup: Avoid a format truncation. Ensure that output buffers for the osp_convert_inout function have sufficient space for additional data such as brackets and ports. ASTERISK-28804 Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b	6 years ago
Pirmin Walthert	6b2d945174	app.c: make sure that no non-async-signal-safe syscalls are used after fork before exec Posix does only allow async-signal-safe syscalls after fork before exec. As asterisk ignores this, functions like TrySystem or System sometimes end up in a deadlocked child process. The patch prevents the use of non-async-signal-safe syscalls. ASTERISK-28776 Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e	6 years ago
George Joseph	7fbfbe7da0	streams: Fix one memory leak and one formats ref issue ast_stream_topology_create_from_format_cap() was setting the stream->formats directly but not freeing the default formats. This causes a memory leak. * ast_stream_topology_create_from_format_cap() now calls ast_stream_set_formats() which properly cleans up the existing stream formats. When cloning a stream, the source stream's format caps _pointer_ is copied to the new stream and it's reference count bumped. If either stream is set to "removed", this will cause _both_ streams to have their format caps cleared. * ast_stream_clone() now creates a new format caps object and copies the formats from the source stream instead of just copying the pointer. ASTERISK-28870 Change-Id: If697d81c3658eb7baeea6dab413b13423938fb53	6 years ago
Nathan Bruning	f217fcdc62	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	6 years ago
Jaco Kroon	44e5dd288b	Remove #include <sys/cdefs.h> These are not provided by standards, and as a result causes failure to compile on musl. https://wiki.musl-libc.org/faq.html#Q:-When-compiling-something-against-musl,-I-get-error-messages-about-%3Ccode%3Esys/cdefs.h%3C/code%3E Change-Id: I6a357cefd106c72cfecafd898638f6b5692c2e05	6 years ago
Guido Falsi	c831f03273	pjproject: Remove bashism from configure.m4 script The configure.m4 script for pjproject contains some += syntax, which is specific to bash, replacing it with string substitutions makes the script compatible with traditional Bourne shells. ASTERISK-28866 #close Reported-by: Christoph Moench-Tegeder <cmt@FreeBSD.org> Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05	6 years ago
Joshua C. Colp	1cfd30bd8a	res_stir_shaken: Use ast_asprintf for creating file path. Change-Id: Ice5d92ecea2f1101c80487484f48ef98be2f1824	6 years ago
Ben Ford	9acf840f7c	res_stir_shaken: Implemented signature verification. There are a lot of moving parts in this patch, but the focus of it is on the verification of the signature using a public key located at the public key URL provided in the JSON payload. First, we check the database to see if we have already downloaded the key. If so, check to see if it has expired. If it has, redownload from the URL. If we don't have an entry in the database, just go ahead and download the public key. The expiration is tested each time we download the file. After that, read the public key from the file and use it to verify the signature. All sanity checking is done when the payload is first received, so the verification is complete once this point is reached. The XML has also been added since a new config option was added to general (curl_timeout). The maximum amount of time to wait for a download can be configured through this option, with a low value by default. Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c	6 years ago
George Joseph	7baf2c4bf1	app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict The gcc 10 -Wrestrict option was causing "overlap" errors when snprintf was copying one char[256] structure member to another char[256] member in the same structure. Using ast_alloca instead of declaring the structure inline solves the issue. Here's a link to the "enhancement": https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html We may follow up with a gcc bug report. Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534	6 years ago
Joshua C. Colp	3078a00a6d	pjsip: Increase maximum ICE candidate count. In practice it has been seen that some users come close to our maximum ICE candidate count of 32. In case people have gone over this increases the count to 64, giving ample room. ASTERISK-28859 Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf	6 years ago
Alexander Traud	29070b61f7	core_local: Local calls are always secure. In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support the channel items 'secure_bridge_media' and 'secure_bridge_signaling'. That way, a channel can be forced to use encryption even if not specified in its configuration. However, when the Local Proxy kicks in, for example, in case of a forwarding (SIP status 302), Local Proxy stated it does not know those items. Consequently, such a call could not be proxied how clever your Dialplan was. Because local calls within Asterisk are always secure, Local Proxy accepts such a request now. ASTERISK-22920 Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c	6 years ago
Guido Falsi	e4366308e1	res_rtp_asterisk: Protect access to nochecksums with #ifdef Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. ASTERISK-28852 #close Change-Id: I381718893b80599ab8635f2b594a10c1000d595e	6 years ago
Guido Falsi	97494d8984	core/dns: Add system include required on FreeBSD While testing the latest RC on FreeBSD I noticed this new file fails to build. On FreeBSD inlcuding resolv.h requires sockaddr_in to be defined, and it's defined in netinet/in.h. So I added this include. ASTERISK-28853 #close Change-Id: I6997daf3956e6eb70ab6cb358628d162fad80079	6 years ago
Peter Turczak	3303defd3f	chan_mobile: Add smoother to make SIP/RTP endpoints happy. In contrast to RFC 3551, section 4.2, several SIP/RTP clients misbehave severly (up to crashing). This patch adds another smoother for the audio received via bt. Therefore the audio frames sent to the core will be CHANNEL_FRAME_SIZE. ASTERISK-28832 #close Change-Id: Ic5f9e2f35868ae59cc9356afbd1388b779a1267f	6 years ago
Alexander Traud	26b8c99963	app_fax: SpanDSP headers do not use ast_malloc; ignore that. Since Asterisk 14, app_fax did not compile at all because Asterisk requires that not malloc but ast_malloc is used everywhere. However, the system headers of SpanDSP use malloc. Because we cannot (and do not need to) change system headers, let us ignore this. ASTERISK-28848 Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546	6 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
sungtae kim	9ad3d2829c	res_ari_channels: Fixed endpoint 80 characters limit Fixed it to copy the entire string from the requested endpoint body except tech-prefix. ASTERISK-28847 Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25	6 years ago
Joshua C. Colp	e56f4de7e6	fax: Fix crashes in PJSIP re-negotiation scenarios. This change fixes a few re-negotiation issues uncovered with fax. 1. The fax support uses its own mechanism for re-negotiation by conveying T.38 information in its own frames. The new support for re-negotiating when adding/removing/changing streams was also being triggered for this causing multiple re-INVITEs. The new support will no longer trigger when transitioning between fax. 2. In off-nominal re-negotiation cases it was possible for some state information to be left over and used by the next re-negotiation. This is now cleared. ASTERISK-28811 ASTERISK-28839 Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2	6 years ago
DanielYK	9f117ac9ef	res_pjsip: Fixed format of IPv6 addresses for external media addresses ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	6 years ago
Alexander Traud	52f07176b6	chan_sip: externhost/externaddr with non-default TCP/TLS ports. ASTERISK-28372 Reported by: Anton Satskiy ASTERISK-24428 Reported by: sstream Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c	6 years ago
Alexander Traud	abf4d74384	cdr_odbc: Sync load- and build-time deps. MODULEINFO is checked while buidling/linking the module. AST_MODULE_INFO is checked while loading/running the module. ASTERISK-28838 Change-Id: I55dc05ce19552d0415c9045021b42bd82ef44e52	6 years ago
Joshua C. Colp	6cfc6ff53c	confbridge: Add support for disabling text messaging. When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	6 years ago
Alexander Traud	191f136260	res_pjsip_refer: Add build-time dependency. ASTERISK-28838 Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25	6 years ago
Alexander Traud	5c2b8fdeca	app_getcpeid: Add build-time dependency. ASTERISK-28838 Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96	6 years ago
Alexander Traud	008f46bf1e	res_pjsip: Sync load- and build-time deps. MODULEINFO is checked while buidling/linking the module. AST_MODULE_INFO is checked while loading/running the module. ASTERISK-28838 Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042	6 years ago
Alexander Traud	e2affa3b0a	curl: Add build-time dependency. ASTERISK-28838 Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52	6 years ago
Alexander Traud	f1135b453b	res_pjsip: Add build-time dependency. ASTERISK-28838 Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074	6 years ago
Alexander Traud	966acc6251	pjproject_bundled: Honor --without-pjproject. ASTERISK-28837 Change-Id: Id057324912a3cfe6f50af372675626bb515907d9	6 years ago
Pirmin Walthert	d50fd0acc0	res_rtp_asterisk: Resolve loop when receive buffer is flushed When the receive buffer was flushed by a received packet while it already contained a packet with the same sequence number, Asterisk never left the while loop which tried to order the packets. This change makes it so if the packet is in the receive buffer it is retrieved and freed allowing the buffer to empty. ASTERISK-28827 Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3	6 years ago
Alexander Traud	a107e85b2e	install_prereq: Add libcap for high bits in DiffServ/ToS. works automatically; see Mantis 7047 (now ASTERISK-6863) Change-Id: I27d2c109180bd857b6757fd532de48eddb78aee6	6 years ago
Alexander Traud	4d0ab620be	chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. ASTERISK-27195 Reported by: Joshua Roys Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de	6 years ago
Alexander Traud	4ef5ba58f5	BuildSystem: Only if found LibPRI, check its optional parts. Change-Id: If8445f899ee4ce0c606c484943d4ce0c8e43b5da	6 years ago
Pirmin Walthert	ca032d1e2e	res_rtp_asterisk: Free payload when error on insertion to data buffer When the ast_data_buffer_put rejects to add a packet, for example because the buffer already contains a packet with the same sequence number, the payload will never be freed, resulting in a memory leak. The data buffer will now return an error if this situation occurs allowing the caller to free the payload. The res_rtp_asterisk module has also been updated to do this. ASTERISK-28826 Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1	6 years ago
Alexander Traud	ef580f96e7	BuildSystem: Only if found external PJProject, check its optional parts. Change-Id: I11d5693d25c166c99d8cebffc16184d58f6362de	6 years ago
bernard merindol	7db03e12a7	res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0 When the first DTMF receive in RF2833 codec have TimeStamp at 0 is not processed. ASTERISK-28812 Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504	6 years ago
Alexander Traud	611529fa52	res_stir_shaken: Do not build without OpenSSL. Change-Id: Idba5151a3079f9dcc0076d635422c5df5845114f	6 years ago
Alexander Traud	27de0c9700	res_audiosocket: Avoid Sometimes-uninitialized Warning with Clang. Change-Id: I40c014c2cb88e943cf6f1b99a08c7c885e855b3a	6 years ago
Jean Aunis	de66713fd5	func_volume: Accept decimal number as argument Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	6 years ago
Jaco Kroon	2b80e5f5da	res_rtp_asterisk: iterate all local addresses looking to populate ICE. By using pjproject to give us a list of candidates, and then filtering, if the host has >32 addresses configured, then it is possible that we end up filtering out all 32 of those, and ending up with no candidates at all. Instead, get getifaddrs (which pjsip is using underlying anyway) to retrieve all local addresses, and iterate those, adding the first 32 addresses not excluded by the ICE ACL. In our setup at any point in time We've got between 6 and 328 addresses on any given system. The lower limit is the lower limit but the upper limit is growing on a near daily basis currently. Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Alexander Traud	3431949a52	pjproject_bundled: Repair ./configure --with-ssl without ARG. ASTERISK-28758 Reported by: Patrick Wakano Reported by: Dmitriy Serov Change-Id: Ifb6b85c559d116739af00bc48d1f547caa85efac	6 years ago
Jaco Kroon	1cf569ba2b	res_pjsip: document legal dtls_verify endpoint options. Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b	6 years ago
Alexander Traud	610e058189	BuildSystem: Search for Python/C API when possibly needed only. The Python/C API is used only if the Test Framework was enabled in Asterisk 'make menuselect'. The Test Framework is available only if the Developer Mode was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API is used only if the PJProject was found and not disabled in Asterisk; the user did not go for './configure --without-pjproject'. Furthermore, because version 2 of that Python/C API is required (currently) and because some platforms do not offer a generic version 2, the script searches for 2.7 explicitly as well. To avoid version mismatch between the Python/C API and the Python environment, the script searches for the latter in the same versions, in the same the order as well. Because this Python/C API is just for (some) Asterisk contributors, the script also goes for the Python 3 environment as a last resort for all other Asterisk users. This allows 'make full' even on minimal installations of Ubuntu 18.04 LTS and newer. Because the Python/C API is Asterisk contributor specific, the Python packages are removed from the script './contrib/scripts/install_prereq' as this script is intended for Asterisk users. Asterisk contributors have to install much more packages in any case, like: sudo apt install autoconf automake git git-review python2.7-dev ASTERISK-28824 ASTERISK-27717 Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876	6 years ago

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