5457 Commits (d1fb397cfc1be59e5e0a3508502b407a6e1c45a3)

Author	SHA1	Message	Date
Naveen Albert	d1fb397cfc	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	2 years ago
Matthew Fredrickson	45da3ff9fa	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	2 years ago
Sean Bright	8d87d403bc	res_pjsip_header_funcs.c: Check URI parameter length before copying. ... Fixes #477	2 years ago
Sean Bright	83636e4b92	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.	2 years ago
Sean Bright	611010d67d	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	2 years ago
Sean Bright	c19470497f	resource_channels.c: Explicit codec request when creating UnicastRTP. ... Fixes #394	2 years ago
Sean Bright	c040179fcf	doc: Update IP Quality of Service links. ... Fixes #328	2 years ago
George Joseph	f309ffad3d	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
Holger Hans Peter Freyther	69590ba33e	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.	2 years ago
Brad Smith	6ec59e1e04	res_rtp_asterisk.c: Fix runtime issue with LibreSSL ... The module will fail to load. Use proper function DTLS_method() with LibreSSL.	2 years ago
Naveen Albert	b94f8bb216	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	2 years ago
Mike Bradeen	f666dd0dd8	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.	2 years ago
Sean Bright	99527745eb	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. ... Fixes #386	2 years ago
Sean Bright	8283aa40a0	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. ... Fixes #376	2 years ago
Sean Bright	8a27d7ef89	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.	2 years ago
George Joseph	c32d090e77	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	2 years ago
sungtae kim	ddb5c377fd	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
Mike Bradeen	2694792e13	res_stasis: signal when new command is queued ... res_statsis's app loop sleeps for up to .2s waiting on input to a channel before re-checking the command queue. This can cause delays between channel setup and bridge. This change is to send a SIGURG on the sleeping thread when a new command is enqueued. This exits the sleeping thread out of the ast_waitfor() call triggering the new command being processed on the channel immediately. Resolves: #362 UserNote: Call setup times should be significantly improved when using ARI.	2 years ago
Holger Hans Peter Freyther	b99606955e	ari/stasis: Indicate progress before playback on a bridge ... Make it possible to start a playback and the calling party to receive audio on a bridge before the call is connected. Model the implementation after play_on_channel and deliver a AST_CONTROL_PROGRESS before starting the playback. For a PJSIP channel this will result in sending a SIP 183 Session Progress.	2 years ago
Mike Bradeen	2291f196c5	res_pjsip: update qualify_timeout documentation with DNS note ... The documentation on qualify_timeout does not explicitly state that the timeout includes any time required to perform any needed DNS queries on the endpoint. If the OPTIONS response is delayed due to the DNS query, it can still render an endpoint as Unreachable if the net time is enough for qualify_timeout to expire. Resolves: #352	2 years ago
Mike Bradeen	1885d0677c	res_speech_aeap: add aeap error handling ... res_speech_aeap previously did not register an error handler with aeap, so it was not notified of a disconnect. This resulted in SpeechBackground never exiting upon a websocket disconnect. Resolves: #303	2 years ago
Tinet-mucw	671eeeca24	res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes. ... From the gdb information, ast_websocket_read reads a message successfully, then transport_read is called in the serializer. During execution of pjsip_transport_down, ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop. After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages. This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue. In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop. Resolves: asterisk#299	2 years ago
Vitezslav Novy	8079e5eec4	res_rtp_asterisk: fix wrong counter management in ioqueue objects ... In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased which prevents unused ICE TURN threads from being removed. Resolves: #301	2 years ago
George Joseph	6cc101d886	res_pjsip_pubsub: Add body_type to test_handler for unit tests ... The ast_sip_subscription_handler "test_handler" used for the unit tests didn't set "body_type" so the NULL value was causing a SEGV in build_subscription_tree(). It's now set to "". Resolves: #335	2 years ago
Sean Bright	8cc92b2638	res_stasis_recording.c: Save recording state when unmuted. ... Fixes #322	2 years ago
Mike Bradeen	62541787f2	res_speech_aeap: check for null format on response ... * Fixed issue in res_speech_aeap when unable to provide an input format to check against.	2 years ago
Bastian Triller	468df4a12d	res_pjsip_session: Send Session Interval too small response ... Handle session interval lower than endpoint's configured minimum timer when sending first answer. Timer setting is checked during this step and needs to handled appropriately. Before this change, no response was sent at all. After this change a response with 422 Session Interval too small is sent to UAC.	2 years ago
MikeNaso	eabf036f3d	res_pjsip.c: Set contact_user on incoming call local Contact header ... If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls. Resolves: #226	2 years ago
Sean Bright	3b806a3303	extconfig: Allow explicit DB result set ordering to be disabled. ... Added a new boolean configuration flag - `order_multi_row_results_by_initial_column` - to both res_pgsql.conf and res_config_odbc.conf that allows the administrator to disable the explicit `ORDER BY` that was previously being added to all generated SQL statements that returned multiple rows. Fixes: #179	2 years ago
Naveen Albert	00070bc6bc	res_pjsip_header_funcs: Make prefix argument optional. ... The documentation for PJSIP_HEADERS claims that prefix is optional, but in the code it is actually not. However, there is no inherent reason for this, as users may want to retrieve all header names, not just those beginning with a certain prefix. This makes the prefix optional for this function, simply fetching all header names if not specified. As a result, the documentation is now correct. Resolves: #230 UserNote: The prefix argument to PJSIP_HEADERS is now optional. If not specified, all header names will be returned.	2 years ago
George Joseph	c3c82441a2	Prepare master for Asterisk 22	2 years ago
Maximilian Fridrich	51a7b18038	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
Holger Hans Peter Freyther	f335da6b74	res_prometheus: Do not generate broken metrics ... In 8d6fdf9c3a invisible bridges were skipped but that lead to producing metrics with no name and no help. Keep track of the number of metrics configured and then only emit these. Add a basic testcase that verifies that there is no '(NULL)' in the output. ASTERISK-30474	2 years ago
Sean Bright	c52b4ce11c	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
Sean Bright	fe467d595c	res_geolocation: Ensure required 'location_info' is present. ... Fixes #189	2 years ago
zhengsh	d3c4f93ca6	res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. ... From the gdb information, it was found that when calling __ast_free, the size of the allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb, it is found to be 1. Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid, which is outside the protection of the rtp_instance lock. However, ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses rtp->themssrc_valid within the protection of the rtp_instance lock. This can lead to the possibility that the value of rtp->themssrc_valid used in the call to ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used within ast_rtcp_generate_report(). Resolves: asterisk#63	2 years ago
Mike Bradeen	e84fe59cb2	res_musiconhold: avoid moh state access on unlocked chan ... Move channel unlock to after moh state access to avoid potential unlocked access to state. Resolves: #133	2 years ago
George Joseph	b974a8f9eb	rest-api: Ran make ari stubs to fix resource_endpoints inconsistency	2 years ago
Sean Bright	153095dd91	res_pjsip_rfc3326: Prefer Q.850 cause code over SIP. ... Resolves: #116	2 years ago
Ben Ford	61f37faf6d	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	642b84c9c9	pjsip_transport_events.c: Use %zu printf specifier for size_t. ... Partially resolves #143.	2 years ago
Sean Bright	7e1f0b8e1f	res_crypto.c: Gracefully handle potential key filename truncation. ... Partially resolves #143.	2 years ago
zhou_jiajian	9b8311f9a6	res_fax_spandsp.c: Clean up a spaces/tabs issue	2 years ago
Sean Bright	b4d2e32831	res_crypto.c: Avoid using the non-portable ALLPERMS macro. ... ALLPERMS is not POSIX and it's trivial enough to not jump through autoconf hoops to check for it. Fixes #149.	2 years ago
Jiajian Zhou	73c5967c2d	AMI: Add parking position parameter to Park action ... Add a parking space extension parameter (ParkingSpace) to the Park action. Park action will attempt to park the call to that extension. If the extension is already in use, then execution will continue at the next priority. UserNote: New ParkingSpace parameter has been added to AMI action Park.	2 years ago
Naveen Albert	ce7a72d7e2	res_musiconhold: Add option to loop last file. ... Adds the loop_last option to res_musiconhold, which allows the last audio file in the directory to be looped perpetually once reached, rather than circling back to the beginning again. Resolves: #122 ASTERISK-30462 UserNote: The loop_last option in musiconhold.conf now allows the last file in the directory to be looped once reached.	2 years ago
Joe Searle	8462154a03	res_stasis.c: Add new type 'sdp_label' for bridge creation. ... Add new type 'sdp_label' when creating a bridge using the ARI. This will add labels to the SDP for each stream, the label is set to the corresponding channel id. Resolves: #91 UserNote: When creating a bridge using the ARI the 'type' argument now accepts a new value 'sdp_label' which will configure the bridge to add labels for each stream in the SDP with the corresponding channel id.	2 years ago
InterLinked1	659f2aae3a	res_pjsip_pubsub: Add new pubsub module capabilities. (#82) ... The existing res_pjsip_pubsub APIs are somewhat limited in what they can do. This adds a few API extensions that make it possible for PJSIP pubsub modules to implement richer features than is currently possible. * Allow pubsub modules to get a handle to pjsip_rx_data on subscription * Allow pubsub modules to run a callback when a subscription is renewed * Allow pubsub modules to run a callback for outgoing NOTIFYs, with a handle to the tdata, so that modules can append their own headers to the NOTIFYs This change does not add any features directly, but makes possible several new features that will be added in future changes. Resolves: #81 ASTERISK-30485 #close Master-Only: True	2 years ago
Sean Bright	ae6b56e357	res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#47) ... The functionality we are interested in is present only in pjsip 2.13 and newer. Resolves: #45	2 years ago
zhengsh	c8ce2c705d	res_sorcery_memory_cache.c: Fix memory leak (#56) ... Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue. Resolves: #55 ASTERISK-30429	2 years ago