5488 Commits (4539664c52d8e65c1a847d7fbaf6375b4794de09)

Author	SHA1	Message	Date
George Joseph	4539664c52	stir_shaken: CRL fixes and a new CLI command ... * Fixed a bug in crypto_show_cli_store that was causing asterisk to crash if there were certificate revocation lists in the verification certificate store. We're also now prefixing certificates with "Cert:" and CRLs with "CRL:" to distinguish them in the list. * Added 'untrusted_cert_file' and 'untrusted_cert_path' options to both verification and profile objects. If you have CRLs that are signed by a different CA than the incoming X5U certificate (indirect CRL), you'll need to provide the certificate of the CRL signer here. Thse will show up as 'Untrusted" when showing the verification or profile objects. * Fixed loading of crl_path. The OpenSSL API we were using to load CRLs won't actually load them from a directory, only a file. We now scan the directory ourselves and load the files one-by-one. * Fixed the verification flags being set on the certificate store. - Removed the CRL_CHECK_ALL flag as this was causing all certificates to be checked for CRL extensions and failing to verify the cert if there was none. This basically caused all certs to fail when a CRL was provided via crl_file or crl_path. - Added the EXTENDED_CRL_SUPPORT flag as it is required to handle indirect CRLs. * Added a new CLI command... `stir_shaken verify certificate_file <certificate_file> [ <profile> ]` which will assist troubleshooting certificate problems by allowing the user to manually verify a certificate file against either the global verification certificate store or the store for a specific profile. * Updated the XML documentation and the sample config file. Resolves: #809 (cherry picked from commit d7343d2ba7)	1 year ago
George Joseph	43ca5c0bf9	res_pjsip_config_wizard.c: Refactor load process ... The way we have been initializing the config wizard prevented it from registering its objects if res_pjsip happened to load before it. * We now use the object_type_registered sorcery observer to kick things off instead of the wizard_mapped observer. * The load_module function now checks if res_pjsip has been loaded already and if it was it fires the proper observers so the objects load correctly. Resolves: #816 UserNote: The res_pjsip_config_wizard.so module can now be reloaded. (cherry picked from commit 7b99462959)	1 year ago
George Joseph	a15050650a	res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback ... The ub_result pointer passed to unbound_resolver_callback by libunbound can be NULL if the query was for something malformed like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result and return instead of crashing with a SEGV. This causes pjproject to simply cancel the transaction with a "No answer record in the DNS response" error. The existing "off nominal" unit test was also updated to check this condition. Although not necessary for this fix, we also made ast_dns_resolver_completed() tolerant of a NULL result. Resolves: GHSA-v428-g3cw-7hv9	2 years ago
Igor Goncharovsky	19bbcf0bbf	res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS() ... When using the PJSIP_DIAL_CONTACTS() function for use in the Dial() command, the contacts are returned in text form, so the input to the path_outgoing_request() function is a contact value of NULL. The issue was reported in ASTERISK-28211, but was not actually fixed in ASTERISK-30100. This fix brings back the code that was previously removed and adds code to search for a contact to extract the path value from it. (cherry picked from commit 606df57b64)	2 years ago
Mike Bradeen	e86bd1d97d	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 (cherry picked from commit d59f1a0c44)	2 years ago
George Joseph	831219d74d	security_agreement.c: Always add the Require and Proxy-Require headers ... The `Require: mediasec` and `Proxy-Require: mediasec` headers need to be sent whenever we send `Security-Client` or `Security-Verify` headers but the logic to do that was only in add_security_headers() in res_pjsip_outbound_register. So while we were sending them on REGISTER requests, we weren't sending them on INVITE requests. This commit moves the logic to send the two headers out of res_pjsip_outbound_register:add_security_headers() and into security_agreement:ast_sip_add_security_headers(). This way they're always sent when we send `Security-Client` or `Security-Verify`. Resolves: #789 (cherry picked from commit a45e5ebf8d)	2 years ago
Sean Bright	5f8f4a6852	xml.c: Update deprecated libxml2 API usage. ... Two functions are deprecated as of libxml2 2.12: * xmlSubstituteEntitiesDefault * xmlParseMemory So we update those with supported API. Additionally, `res_calendar_caldav` has been updated to use libxml2's xmlreader API instead of the SAX2 API which has always felt a little hacky (see deleted comment block in `res_calendar_caldav.c`). The xmlreader API has been around since libxml2 2.5.0 which was released in 2003. Fixes #725 (cherry picked from commit 21e3f84e56)	2 years ago
Mike Bradeen	728dbdccd1	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 (cherry picked from commit 182ea91fc5)	2 years ago
George Joseph	741183d7ac	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	2 years ago
Fabrice Fontaine	59aac533c8	res/stasis/control.c: include signal.h ... Include signal.h to avoid the following build failure with uclibc-ng raised since 2694792e13: stasis/control.c: In function 'exec_command_on_condition': stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration] 313 | pthread_kill(control->control_thread, SIGURG); | ^~~~~~~~~~~~ | pthread_yield stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function) 313 | pthread_kill(control->control_thread, SIGURG); | ^~~~~~ cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21 Fixes: #729 (cherry picked from commit d452db3853)	2 years ago
Naveen Albert	d46261b5e4	res_pjsip_logger: Preserve logging state on reloads. ... Currently, reloading res_pjsip will cause logging to be disabled. This is because logging can also be controlled via the debug option in pjsip.conf and this defaults to "no". To improve this, logging is no longer disabled on reloads if logging had not been previously enabled using the debug option from the config. This ensures that logging enabled from the CLI will persist through a reload. ASTERISK-29912 #close Resolves: #246 UserNote: Issuing "pjsip reload" will no longer disable logging if it was previously enabled from the CLI. (cherry picked from commit 001176c239)	2 years ago
Henrik Liljedahl	e27f88ed03	res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting. ... First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation. Fixes #710 (cherry picked from commit 9cf8678112)	2 years ago
George Joseph	824b4ce855	stir_shaken: Fix memory leak, typo in config, tn canonicalization ... * Fixed possible memory leak in tn_config:tn_get_etn() where we weren't releasing etn if tn or eprofile were null. * We now canonicalize TNs before using them for lookups or adding them to Identity headers. * Fixed a typo in stir_shaken.conf.sample. Resolves: #716 (cherry picked from commit 9c1b98e577)	2 years ago
Sperl Viktor	68a4912825	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 (cherry picked from commit 7c3e2cfda9)	2 years ago
George Joseph	6389c743a9	res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2) ... * OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use X509_get_pubkey. The difference is that X509_get_pubkey requires the caller to free the EVP_PKEY themselves so we now let RAII_VAR do that. * OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now wrap it in an ao2 object. * OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all the certs from an X509_STORE and there's no easy way to polyfill it so the CLI commands that list profiles will show a "not supported" message instead of listing the certs in a store. Resolves: #676 (cherry picked from commit af6002564b)	2 years ago
George Joseph	b5865aef32	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org. (cherry picked from commit 3fb9d89586)	2 years ago
Sperl Viktor	bd3c3bf821	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 (cherry picked from commit 136bd5c892)	2 years ago
Joshua Elson	855ed6c168	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 (cherry picked from commit 3d40d34271)	2 years ago
Martin Tomec	378ce2d87f	res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA ... There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip. Fixes: #579 UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip. (cherry picked from commit f3de77f91f)	2 years ago
Martin Nystroem	2c5a1b44f6	res_ari.c: Add additional output to ARI requests when debug is enabled ... When ARI debug is enabled the logs will now output http method and the uri. Fixes: #666 (cherry picked from commit 15f68a836e)	2 years ago
Holger Hans Peter Freyther	081240cf9a	res_prometheus: Fix duplicate output of metric and help text ... The prometheus exposition format requires each line to be unique[1]. This is handled by struct prometheus_metric having a list of children that is managed when registering a metric. In case the scrape callback is used, it is the responsibility of the implementation to handle this correctly. Originally the bridge callback didn't handle NULL snapshots, the crash fix lead to NULL metrics, and fixing that lead to duplicates. The original code assumed that snapshots are not NULL and then relied on "if (i > 0)" to establish the parent/children relationship between metrics of the same class. This is not workerable as the first bridge might be invisible/lacks a snapshot. Fix this by keeping a separate array of the first metric by class. Instead of relying on the index of the bridge, check whether the array has an entry. Use that array for the output. Add a test case that verifies that the help text is not duplicated. Resolves: #642 [1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting (cherry picked from commit d11eba69a6)	2 years ago
Naveen Albert	2bd3d00412	res_parking: Fail gracefully if parking lot is full. ... Currently, if a parking lot is full, bridge setup returns -1, causing dialplan execution to terminate without TryExec. However, such failures should be handled more gracefully, the same way they are on other paths, as indicated by the module's author, here: http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html Now, callers will hear the parking failure announcement, and dialplan will continue, which is consistent with existing failure modes. Resolves: #624 (cherry picked from commit b05865d8e9)	2 years ago
Maximilian Fridrich	4c10d95529	res_pjsip_session: Reset pending_media_state->read_callbacks ... In handle_negotiated_sdp the pending_media_state->read_callbacks must be reset before they are added in the SDP handlers in handle_negotiated_sdp_session_media. Otherwise, old callbacks for removed streams and file descriptors could be added to the channel and Asterisk would poll on non-existing file descriptors. Resolves: #611 (cherry picked from commit 3d71b494d8)	2 years ago
Sean Bright	c89e91c8d0	res_monitor.c: Don't emit a warning about 'X' being unrecognized. ... Code was added in 030f7d41 to warn if an unrecognized option was passed to an application, but code in Monitor was taking advantage of the fact that the application would silently accept an invalid option. We now recognize the invalid option but we don't do anything if it's set. Fixes #639 (cherry picked from commit 6938b7f962)	2 years ago
George Joseph	d7597e5148	res_pjsip_stir_shaken.c: Add checks for missing parameters ... * Added checks for missing session, session->channel and rdata in stir_shaken_incoming_request. * Added checks for missing session, session->channel and tdata in stir_shaken_outgoing_request. Resolves: #645	2 years ago
George Joseph	44563b5043	attestation_config.c: Use ast_free instead of ast_std_free ... In as_check_common_config, we were calling ast_std_free on raw_key but raw_key was allocated with ast_malloc so it should be freed with ast_free. Resolves: #636 (cherry picked from commit 267348bee3)	2 years ago
George Joseph	fd27df9479	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. (cherry picked from commit e6c7f1aee0)	2 years ago
romryz	a494b61fe2	res_rtp_asterisk.c: Correct coefficient in MOS calculation. ... Media Experience Score relies on incorrect pseudo_mos variable calculation. According to forming an opinion section of the documentation, calculation relies on ITU-T G.107 standard: https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion ITU-T G.107 Annex B suggests to calculate MOS with a coefficient "seven times ten to the power of negative six", 7 * 10^(-6). which would mean 6 digits after the decimal point. Current implementation has 7 digits after the decimal point, which downrates the calls. Fixes: #597 (cherry picked from commit 335b925fac)	2 years ago
George Joseph	6df5fbee65	Reduce startup/shutdown verbose logging ... When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582 (cherry picked from commit a433ed0d5a)	2 years ago
Sean Bright	f74eafcf65	res_pjsip_t38.c: Permit IPv6 SDP connection addresses. ... The existing code prevented IPv6 addresses from being properly parsed. Fixes #558 (cherry picked from commit 103a6dc987)	2 years ago
Sean Bright	4d09fb191c	res_pjsip_session.c: Correctly format SDP connection addresses. ... Resolves a regression identified by @justinludwig involving the rendering of IPv6 addresses in outgoing SDP. Also updates `media_address` on PJSIP endpoints so that if we are able to parse the configured value as an IP we store it in a format that we can directly use later. Based on my reading of the code it appeared that one could configure `media_address` as: ``` [foo] type = endpoint ... media_address = [2001:db8::] ``` And that value would be blindly copied into the outgoing SDP without regard to its format. Fixes #541 (cherry picked from commit 0fdaf7fc80)	2 years ago
Naveen Albert	309ec5e762	res_calendar_icalendar: Print iCalendar error on parsing failure. ... If libical fails to parse a calendar, print the error message it provdes. Resolves: #492 (cherry picked from commit 8eb6a329d6)	2 years ago
George Joseph	a42c5438e9	Revert "core & res_pjsip: Improve topology change handling." ... This reverts commit 315eb551db. Over the past year, we've had several reports of "topology storms" occurring where 2 external facing channels connected by one or more local channels and bridges will get themselves in a state where they continually send each other topology change requests. This usually manifests itself in no-audio calls and a flood of "Exceptionally long queue length" messages. It appears that this commit is the cause so we're reverting it for now until we can determine a more appropriate solution. Resolves: #530 (cherry picked from commit 4715c1b11c)	2 years ago
Maximilian Fridrich	b2eb25a88c	res_pjsip_nat: Fix potential use of uninitialized transport details ... The ast_sip_request_transport_details must be zero initialized, otherwise this could lead to a SEGV. Resolves: #509 (cherry picked from commit 3e069f3274)	2 years ago
Naveen Albert	91127a618f	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430 (cherry picked from commit 8f5581b0d0)	2 years ago
Matthew Fredrickson	37a89d3cee	res_odbc.c: Allow concurrent access to request odbc connections ... There are valid scenarios where res_odbc's connection pool might have some dead or stuck connections while others are healthy (imagine network elements/firewalls/routers silently timing out connections to a single DB and a single IP address, or a heterogeneous connection pool connected to potentially multiple IPs/instances of a replicated DB using a DNS front end for load balancing and one replica fails). In order to time out those unhealthy connections without blocking access to other parts of Asterisk that may attempt access to the connection pool, it would be beneficial to not lock/block access around the entire pool in _ast_odbc_request_obj2 while doing potentially blocking operations on connection pool objects such as the connection_dead() test, odbc_obj_connect(), or by dereferencing a struct odbc_obj for the last time and triggering a odbc_obj_disconnect(). This would facilitate much quicker and concurrent timeout of dead connections via the connection_dead() test, which could block potentially for a long period of time depending on odbc.ini or other odbc connector specific timeout settings. This also would make rapid failover (in the clustered DB scenario) much quicker. This patch changes the locking in _ast_odbc_request_obj2() to not lock around odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to lock around truly shared, non-immutable state like the connection_cnt member and the connections list on struct odbc_class. Fixes: #465 (cherry picked from commit bfac3945f6)	2 years ago
Sean Bright	5988da4ec5	res_pjsip_header_funcs.c: Check URI parameter length before copying. ... Fixes #477 (cherry picked from commit a2f0d99d9d)	2 years ago
Sean Bright	d52c427533	res_rtp_asterisk.c: Update for OpenSSL 3+. ... In 5ac5c2b0 we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence deprecation warnings. This commit switches over to using non-deprecated API. (cherry picked from commit 3859b630a7)	2 years ago
Sean Bright	fe92d09361	res_http_websocket.c: Set hostname on client for certificate validation. ... Additionally add a `assert()` to in the TLS client setup code to ensure that hostname is set when it is supposed to be. Fixes #433 (cherry picked from commit f2961f048d)	2 years ago
Sean Bright	da35b6a244	resource_channels.c: Explicit codec request when creating UnicastRTP. ... Fixes #394 (cherry picked from commit a83c761c95)	2 years ago
Sean Bright	fb7d39db6d	doc: Update IP Quality of Service links. ... Fixes #328 (cherry picked from commit 26918d05f4)	2 years ago
George Joseph	94f931a6d7	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. (cherry picked from commit cd77953172)	2 years ago
Holger Hans Peter Freyther	28f52d35f3	ari: Provide the caller ID RDNIS for the channels ... Provide the caller ID RDNIS when available. This will allow an application to follow the redirect. (cherry picked from commit 157389bc59)	2 years ago
Brad Smith	fb3067dd71	res_rtp_asterisk.c: Fix runtime issue with LibreSSL ... The module will fail to load. Use proper function DTLS_method() with LibreSSL. (cherry picked from commit 65d38c8104)	2 years ago
Naveen Albert	120dbba2b3	res_pjsip: Include cipher limit in config error message. ... If too many ciphers are specified in the PJSIP config, include the maximum number of ciphers that may be specified in the user-facing error message. Resolves: #396 (cherry picked from commit 8d9d0d3738)	2 years ago
Mike Bradeen	0b4427d621	res_speech: allow speech to translate input channel ... * Allow res_speech to translate the input channel if the format is translatable to a format suppored by the speech provider. Resolves: #129 UserNote: res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites. (cherry picked from commit c3e2bff36e)	2 years ago
Sean Bright	ea74f942ce	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. ... Fixes #386 (cherry picked from commit deeb1acffe)	2 years ago
Sean Bright	58f78d2d0e	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. ... Fixes #376 (cherry picked from commit d4e4942cf5)	2 years ago
Sean Bright	93697458a6	pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. ... Per RFC8827: Implementations MUST NOT implement DTLS renegotiation and MUST reject it with a "no_renegotiation" alert if offered. So we disable it when webrtc=yes is set. Fixes #378 UpgradeNote: The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. (cherry picked from commit 9d329da346)	2 years ago
George Joseph	e52175341e	res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown ... Commit f66f77f last year prevents the res_pjsip_exten_state and res_pjsip_mwi modules from unloading due to possible pjproject asserts if the modules are reloaded. A side effect of the implementation is that the taskprocessors these modules use aren't being released. When asterisk is doing a graceful shutdown, it waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all taskprocessors to stop but since those 2 modules don't release theirs, the shutdown hangs for that amount of time. This change allows the modules to be unloaded and their resources to be released when ast_shutdown_final is true. Resolves: #379 (cherry picked from commit 9efc4bdfbc)	2 years ago