4375 Commits (18)

Author	SHA1	Message	Date
George Joseph	e5f0a1467a	asterisk.c: Add option to restrict shell access from remote consoles. ... UserNote: A new asterisk.conf option 'disable_remote_console_shell' has been added that, when set, will prevent remote consoles from executing shell commands using the '!' prefix. Resolves: #GHSA-c7p6-7mvq-8jq2	1 week ago
Allan Nathanson	99221c81ed	dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin ... The dnsmgr_refresh() function checks to see if the IP address associated with a name/service has changed. The gotcha is that the ast_get_ip_or_srv() function only returns the first IP address returned by the DNS query. If there are multiple IPs associated with the name and the returned order is not consistent (e.g. with DNS round-robin) then the other IP addresses are not included in the comparison and the entry is flagged as changed even though the IP is still valid. Updated the code to check all IP addresses and flag a change only if the original IP is no longer valid. Resolves: #924	8 months ago
Sean Bright	c80ce750bb	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. ... Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922	8 months ago
Mike Bradeen	cf5a6435c2	res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch ... When Asterisk sends an offer to Bob that includes 48K and 8K codecs with matching 4733 offers, Bob may want to use the 48K audio codec but can not accept 48K digits and so negotiates for a mixed set. Asterisk will now check Bob's offer to make sure Bob has indicated this is acceptible and if not, will use Bob's preference. Fixes: #847	9 months ago
Ben Ford	027127246e	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	10 months ago
Mike Bradeen	ac9c510d99	res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits ... After change made in 624f509 to add support for non 8K RFC 4733/2833 digits, Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of the negotiated codec(s). This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec. A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to be re-written to allow for these scenarios. Fixes: #776	11 months ago
Sean Bright	0bcee9de5d	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. ... Fixes #785	11 months ago
Sean Bright	9942ebae86	file.h: Rename function argument to avoid C++ keyword clash. ... Fixes #744	12 months ago
George Joseph	463cf883d9	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
Mike Bradeen	6bf66b82d7	rtp_engine: add support for multirate RFC2833 digits ... Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs. Asterisk currently treats RFC2833 Digits as a single rtp payload type with a fixed bitrate of 8K. This change would expand that to 8, 16, 24 and 32K. This requires checking the offered rtp types for any of these bitrates and then adding an offer for each (if configured for RFC2833.) DTMF generation must also be changed in order to look at the current outbound codec in order to generate appropriately timed rtp. For cases where no outgoing audio has yet been sent prior to digit generation, Asterisk now has a concept of a 'preferred' codec based on offer order. On inbound calls Asterisk will mimic the payload types of the RFC2833 digits. On outbound calls Asterisk will choose the next free payload types starting with 101. UserNote: No change in configuration is required in order to enable this feature. Endpoints configured to use RFC2833 will automatically have this enabled. If the endpoint does not support this, it should not include it in the SDP offer/response. Resolves: #699	1 year ago
Naveen Albert	23f5ce69fd	logger: Add unique verbose prefixes for levels 5-10. ... Add unique verbose prefixes for levels higher than 4, so that these can be visually differentiated from each other. Resolves: #721	1 year ago
Naveen Albert	c276ae11e0	loader.c: Allow dependent modules to be unloaded recursively. ... Because of the (often recursive) nature of module dependencies in Asterisk, hot swapping a module on the fly is cumbersome if a module is depended on by other modules. Currently, dependencies must be popped manually by unloading dependents, unloading the module of interest, and then loading modules again in reverse order. To make this easier, the ability to do this recursively in certain circumstances has been added, as an optional extension to the "module refresh" command. If requested, Asterisk will check if a module that has a positive usecount could be unloaded safely if anything recursively dependent on it were unloaded. If so, it will go ahead and unload all these modules and load them back again. This makes hot swapping modules that provide dependencies much easier. Resolves: #474 UserNote: In certain circumstances, modules with dependency relations can have their dependents automatically recursively unloaded and loaded again using the "module refresh" CLI command or the ModuleLoad AMI command.	1 year ago
George Joseph	f9a1e3095c	tcptls/iostream: Add support for setting SNI on client TLS connections ... If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello.	1 year ago
Naveen Albert	6bd0b67081	callerid.c: Parse previously ignored Caller ID parameters. ... Commit f2f397c1a8 previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681	1 year ago
George Joseph	65018e8ebf	logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT ... If you're tracing a large function that may call another function multiple times in different circumstances, it can be difficult to see from the trace output exactly which location that function was called from. There's no good way to automatically determine the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT simply print out a trace line before and after the call. The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is that SCOPE_CALL ignores the function's return value (if any) where SCOPE_CALL_WITH_RESULT allows you to specify the type of the function's return value so it can be assigned to a variable. SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT and the "int" return type.	1 year ago
Sperl Viktor	62bee37d0d	res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address. ... Add a new identify_by option to res_pjsip_endpoint_identifier_ip called 'transport' this matches endpoints based on the bound ip address (local) instead of the 'ip' option, which matches on the source ip address (remote). UserNote: set identify_by=transport for the pjsip endpoint. Then use the existing 'match' option and the new 'transport' option of the identify. Fixes: #672	1 year ago
Sperl Viktor	83f1317eb4	res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI ... Add ability to match against PJSIP request URI. UserNote: this new feature let users match endpoints based on the indound SIP requests' URI. To do so, add 'request_uri' to the endpoint's 'identify_by' option. The 'match_request_uri' option of the identify can be an exact match for the entire request uri, or a regular expression (between slashes). It's quite similar to the header identifer. Fixes: #599	1 year ago
Joshua Elson	f61e1d902b	Implement Configurable TCP Keepalive Settings in PJSIP Transports ... This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary. Fixes #657	1 year ago
George Joseph	181edcc3a3	Stir/Shaken Refactor ... Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.	1 year ago
Sean Bright	164001b05c	strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. ... If a dynamic string is created with an initial length of 0, `ast_str_buffer(…)` will return an invalid pointer. This was a secondary discovery when fixing #65.	1 year ago
Naveen Albert	72f4cf6798	configure: Rerun bootstrap on modern platform. ... The last time configure was run, it was run on a system that did not enable -std=gnu11 by default, which meant that the restrict qualifier would not be recognized on certain platforms. This regenerates the configure files from running bootstrap.sh, so that these should be recognized on all supported platforms. Resolves: #586	1 year ago
Joshua C. Colp	5e09e2afeb	utils: Make behavior of ast_strsep* match strsep. ... Given the scenario of passing an empty string to the ast_strsep functions the functions would return NULL instead of an empty string. This is counter to how strsep itself works. This change alters the behavior of the functions to match that of strsep. Fixes: #565	1 year ago
Brad Smith	75dc596843	main/utils: Simplify the FreeBSD ast_get_tid() handling ... FreeBSD has had kernel threads for 20+ years.	1 year ago
George Joseph	f00e026416	res_rtp_asterisk: Fix regression issues with DTLS client check ... * Since ICE candidates are used for the check and pjproject is required to use ICE, res_rtp_asterisk was failing to compile when pjproject wasn't available. The check is now wrapped with an #ifdef HAVE_PJPROJECT. * The rtp->ice_active_remote_candidates container was being used to check the address on incoming packets but that container doesn't contain peer reflexive candidates discovered during negotiation. This was causing the check to fail where it shouldn't. We now check against pjproject's real_ice->rcand array which will contain those candidates. * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where we weren't zeroing out sin->sin_zero before returning. This was causing ast_sockaddr_cmp() to always return false when one of the inputs was converted from a pj_sockaddr, even if both inputs had the same address and port. Resolves: #500 Resolves: #503 Resolves: #505	1 year ago
Naveen Albert	147f014072	func_channel: Expose previously unsettable options. ... Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442	2 years ago
George Joseph	9a93ce0409	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	2 years ago
sungtae kim	96420f3d48	res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters ... This commit introduces an extension to the endpoint and relevant resource sizes for PJSIP, transitioning from its current 40-character constraint to a more versatile 255-character capacity. This enhancement significantly overcomes limitations related to domain qualification and practical usage, ultimately delivering improved functionality. In addition, it includes adjustments to accommodate the expanded realm size within the ARI, specifically enhancing the maximum realm length. Resolves: #345 UserNote: With this update, the PJSIP realm lengths have been extended to support up to 255 characters. UpgradeNote: As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas.	2 years ago
George Joseph	f74c84e978	logger.h: Add ability to change the prefix on SCOPE_TRACE output ... You can now define the _TRACE_PREFIX_ macro to change the default trace line prefix of "file:line function" to something else. Full documentation in logger.h.	2 years ago
George Joseph	761b143db3	Add libjwt to third-party ... The current STIR/SHAKEN implementation is not currently usable due to encryption issues. Rather than trying to futz with OpenSSL and the the current code, we can take advantage of the existing capabilities of libjwt but we first need to add it to the third-party infrastructure already in place for jansson and pjproject. A few tweaks were also made to the third-party infrastructure as a whole. The jansson "dest" install directory was renamed "dist" to better match convention, and the third-party Makefile was updated to clean all product directories not just the ones currently in use. Resolves: #349	2 years ago
Bastian Triller	6edeb90485	func_json: Fix crashes for some types ... This commit fixes crashes in JSON_DECODE() for types null, true, false and real numbers. In addition it ensures that a path is not deeper than 32 levels. Also allow root object to be an array. Add unit tests for above cases.	2 years ago
George Joseph	04df168656	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS ... Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing.	2 years ago
Mike Bradeen	ff4b5ed951	cel: add publish user event helper ... Add a wrapper function around ast_cel_publish_event that packs event and extras into a blob before publishing Resolves:#330	2 years ago
George Joseph	c8a97d5f8c	file.c: Add ability to search custom dir for sounds ... To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory.	2 years ago
George Joseph	55eca816b1	make_buildopts_h, et. al. Allow adding all cflags to buildopts.h ... The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options.	2 years ago
Maximilian Fridrich	57f77e8218	core/ari/pjsip: Add refer mechanism ... This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.	2 years ago
Sean Bright	8e23f3e313	res_pjsip: Enable TLS v1.3 if present. ... Fixes #221 UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying PJSIP library. The bundled version of PJSIP supports TLS v1.3.	2 years ago
George Joseph	d54695914d	app.h: Move declaration of ast_getdata_result before its first use ... The ast_app_getdata() and ast_app_getdata_terminator() declarations in app.h were changed recently to return enum ast_getdata_result (which is how they were defined in app.c). The existing declaration of ast_getdata_result in app.h was about 1000 lines after those functions however so under certain circumstances, a "use before declaration" error was thrown by the compiler. The declaration of the enum was therefore moved to before those functions. Resolves: #200	2 years ago
Ben Ford	23c4d21d1b	res_pjsip_session: Added new function calls to avoid ABI issues. ... Added two new functions (ast_sip_session_get_dialog and ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the pjsip_inv_state respectively from the pjsip_inv_session on the ast_sip_session struct. This is due to pjproject adding a new field to the pjsip_inv_session struct that caused crashes when trying to access fields that were no longer where they were expected to be if a module was compiled against a different version of pjproject. Resolves: #145	2 years ago
Sean Bright	a36af23f20	configure: Remove obsolete and deprecated constructs. ... These were uncovered when trying to run `bootstrap.sh` with Autoconf 2.71: * AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS(). * AC_HEADER_TIME is obsolete. * $as_echo is deprecated in favor of AS_ECHO() which requires an update to ax_pthread.m4. Note that the generated artifacts in this commit are from Autoconf 2.69. Resolves #139	2 years ago
George Joseph	acb18c1fc4	build: Fix a few gcc 13 issues ... * gcc 13 is now catching when a function is declared as returning an enum but defined as returning an int or vice versa. Fixed a few in app.h, loader.c, stasis_message.c. * gcc 13 is also now (incorrectly) complaining of dangling pointers when assigning a pointer to a local char array to a char *. Had to change that to an ast_alloca. Resolves: #155	2 years ago
Naveen Albert	8a03ed6877	callerid: Allow specifying timezone for date/time. ... The Caller ID generation routine currently is hardcoded to always use the system time zone. This makes it possible to optionally specify any TZ-format time zone. Resolves: #98 ASTERISK-30330	2 years ago
Maximilian Fridrich	a4cd452246	chan_pjsip: Allow topology/session refreshes in early media state ... With this change, session modifications in the early media state are possible if the SDP was sent reliably and confirmed by a PRACK. For details, see RFC 6337, escpecially section 3.2. Resolves: #73	2 years ago
Sean Bright	3e2a28fc3d	res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76) ... The functionality we are interested in is present only in pjsip 2.13 and newer. Resolves: #45	2 years ago
Mike Bradeen	fa18f2d71e	cel: add local optimization begin event ... The current AST_CEL_LOCAL_OPTIMIZE event is and has been triggered on a local optimization end to serve as a flag indicating the event occurred. This change adds a second AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail. Resolves: #52 UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired. UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.	2 years ago
Sean Bright	6f218514fb	core: Cleanup gerrit and JIRA references. (#40) ... * Remove .gitreview and switch to pulling the main asterisk branch version from configure.ac instead. * Replace references to JIRA with GitHub. * Other minor cleanup found along the way. Resolves: #39	2 years ago
Maximilian Fridrich	ecd5f91125	res_pjsip: mediasec: Add Security-Client headers after 401 ... When using mediasec, requests sent after a 401 must still contain the Security-Client header according to draft-dawes-sipcore-mediasec-parameter. Resolves: #48	2 years ago
Naveen Albert	4e602a1afe	pbx_dundi: Add PJSIP support. ... Adds PJSIP as a supported technology to DUNDi. To facilitate this, we now allow an endpoint to be specified for outgoing PJSIP calls. We also allow users to force a specific channel technology for outgoing SIP-protocol calls. ASTERISK-28109 #close ASTERISK-28233 #close Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf	2 years ago
Naveen Albert	1bbcb98558	res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters. ... The current STIR/SHAKEN signing process is inconsistent with the RFCs in a couple ways that can cause interoperability issues. RFC8225 specifies that the keys must be ordered lexicographically, but currently the fields are simply ordered according to the order in which they were added to the JSON object, which is not compliant with the RFC and can cause issues with some carriers. To fix this, we now leverage libjansson's ability to dump a JSON object sorted by key value, yielding the correct field ordering. Additionally, telephone numbers must have any leading + prefix removed and must not contain characters outside of 0-9, *, and # in order to comply with the RFCs. Numbers are now properly formatted as such. ASTERISK-30407 #close Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2	2 years ago
Mike Bradeen	b2e9419961	res_mixmonitor: MixMonitorMute by MixMonitor ID ... While it is possible to create multiple mixmonitor instances on a channel, it was not previously possible to mute individual instances. This change includes the ability to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via id. This id can be stored as a channel variable using the 'i' MixMonitor option. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor spy-type audiohooks on the channel. This is done via the new audiohook function: ast_audiohook_set_mute_all. ASTERISK-30464 Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98	2 years ago
George Joseph	1ddfb7551a	res_pjsip: Replace invalid UTF-8 sequences in callerid name ... * Added a new function ast_utf8_replace_invalid_chars() to utf8.c that copies a string replacing any invalid UTF-8 sequences with the Unicode specified U+FFFD replacement character. For example: "abc\xffdef" becomes "abc\uFFFDdef". Any UTF-8 compliant implementation will show that character as a � character. * Updated res_pjsip:set_id_from_hdr() to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. * Updated stasis_channels:ast_channel_publish_varset to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. ASTERISK-27830 Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8	2 years ago