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Author	SHA1	Message	Date
Joshua C. Colp	cae348fed1	extension_state: Add new extension state API. ... Extension state to this point has been an API implemented inside the PBX core resulting in its state being intermingled with that of the dialplan. This increased the complexity of the PBX core and made it difficult to enact improvements. This change adds a new separate extension state API which receives updates from the PBX core as configuration changes but maintains its own separate state. The API is also written to fully take advantage of modern APIs in a more selective manner by subscribing each extension state to only the devices it is interested in, ultimately reducing resource consumption during updates. Presence state updates being infrequently done use a single shared subscription that goes through the extension states to find and update ones that the update is applicable to. Legacy API support is provided by reimplementing the API as wrappers over the new extension state API. This improves the legacy API by making it multithreaded, with each callback being individually subscribed. Autohints support is maintained but has been separated out into a self contained new implementation. Synchronous subscription support has also been added to Stasis to remove the overhead of asynchronous publishing when the handling of published messages is small and fast.	3 weeks ago
Alexis Chenard	b7a1eb6b9a	res_pjsip: Add external_signaling_hostname transport option ... Adds a new transport option 'external_signaling_hostname' which allows a hostname or FQDN to be used in SIP Contact and Via headers instead of the automatically determined IP address. This is useful when a remote SIP endpoint requires a fully qualified domain name in these headers. The option is mutually exclusive with 'external_signaling_address' and an error is raised at transport load time if both are set simultaneously. Resolves: #1749 UserNote: A new pjsip.conf transport option 'external_signaling_hostname' has been added. When set, this value will be used in SIP Contact and Via headers instead of the automatically determined IP address. This option is mutually exclusive with 'external_signaling_address'.	3 weeks ago
George Joseph	7c8c112361	WebSocket Enhancements: Proxies and Keepalives for ARI and Media Outbound Websockets. ... See the notes below for high-level descriptions of the new features. * Proxies Outbound/forward HTTP proxies are now supported and configurable in websocket_client.conf. You can specify a host:port plus optional proxy_username and proxy_password. Because WebSockets aren't consistently supported among proxies (specifically passing through UPGRADEs), the CONNECT method is always used to establish a TCP tunnel through the proxy. This is required if a TLS session is to be established with the WebSocket server anyway. It's important to understand that that negotiation with the proxy is ALWAYS unsecured. Once the proxy establishes the tunnel, the TLS session will be negotiated directly with the remote WebSocket server via the tunnel. * Keepalives Both TCP-level and WebSocket PING/PONG keepalives can be configured and are available with either the curl or tcptls client implementations. The TCP keepalives are handled entirely by the operating system and require no resources from Asterisk but by their very nature, they can't traverse proxies. WebSocket PING/PONGs are implemented in the Asterisk websocket code and require a scheduler thread to keep track of them so they're a bit more complicated but they do traverse proxies. Which one is used is completely up to the admin. You could use both. * Other Changes A few changes were needed to res/ari/ari_websockets and res/res_aeap/transport_websocket to add explicit calls to ast_websocket_close. They had been assuming that the websocket session destructor would close the websocket when it unreffed it but the keepalive process now holds a reference so the destructor wouldn't actually run without the call to ast_websocket_close to stop the keepalives. A few new methods were added to tcptls.c to allow switching an existing connection from unsecured to TLS. These were required because the initial connection and handshake with a proxy is always unsecured but then needs to be switched to TLS if required for the remote WebSocket server. There was a bug in sorcery.h where the ast_sorcery_register_uint macro was referencing _stringify (which doesn't exist) instead of _sorcery_stringify. Resolves: #1881 Resolves: #1933 UserNote: Forward/outbound proxies can now be specified for outbound websockets. See the websocket_client.conf.sample file for configuration information. UserNote: TCP-level or WebSocket PING/PONG keepalives can now be enabled on outbound websockets. They can help detect network failures even when a persistent connection is idle. See the websocket_client.conf.sample file for configuration information. DeveloperNote: The addition of the proxy and keepalive configuration parameters pushed the websocket client parameter count over 32. This necessitated changing the size of the ast_ws_client_fields enum from a 32 bit bitfield to a 64-bit bitfield with a corresponding change to the ast_websocket_client structure.	3 weeks ago
Naveen Albert	a41f357a7f	chan_local: Update chan_local references for Local channels. ... chan_local no longer exists since Local channels are built into the core (core_local), but there are still comments which reference it, including in the configs. Update these to avoid confusion. Resolves: #1849	3 weeks ago
Asterisk Development Team	9c6d578ea1	Update for 23.4.1	2 months ago
ThatTotallyRealMyth	15ca862f86	ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory. ... The ast_tsconvert.py script called by ast_loggrabber is now installed in a temporary directory that isn't world readable or writable. Resolves: #GHSA-xgj6-2gc5-5x9c	2 months ago
George Joseph	d803b5370b	chan_unistim.c: Prevent overrun of phone_number field. ... Add a check to key_dial_page() to ensure that dialed digits won't overrun the phone_number field. Resolves: #GHSA-3g56-cgrh-95p5	2 months ago
Mike Bradeen	62a7a7037b	ooh323c: not checking for IE minimum length ... When decoding q.931 encoded calling/called number now checking for length being less than minimum required. Resolves: #GHSA-h5hv-jmgj-92q2	2 months ago
George Joseph	9d198a7d16	res_ari: Ensure read-only users are properly authorized via REST Over WebSocket. ... The REST over WebSocket path now properly prevents non-GET methods from being executed on inbound WebSockets. * The query parameters from the original incoming GET request that caused the upgrade to WebSocket are now passed to all REST requests that come from the client. This ensures that if the client authenticated with a read-only userid using the "api_key" query_string parameter, REST requests coming in over the WebSocket will only be able to execute GETs on resources. The HTTP headers were already passed to the REST requests so if the client had authenticated via an "Authorization" it was properly handled. * New tests have been added to test_ari.c to check that read-only users are properly denied access to resources using non-GET methods. Several memory leaks were also squashed. Resolves: #GHSA-wcvv-g26m-wx5c	2 months ago
George Joseph	6e1f4df744	pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address. ... The filter_on_tx_message() function was using pj_strassign() to save the pointer of the pjproject transport local address to a local pj_str_t variable. That variable was ultimately used to set the Contact header's uri->host and the SDP connection attribute's address again using pj_strassign. pj_strassign() doesn't copy the actual value of the pj_str_t however, it just copies the pointer so if a connection-oriented transport is disconnected before the 200 OK with the SDP is sent, those pointers will be invalid which can cause use-after-free issues. To prevent this, filter_on_tx_message() now uses pj_strdup with the tdata->pool as the backing store to save the local IP address to the local variable. pj_strassign() can then be used safely later on since the tdata will be available for the life of the transaction. Resolves: #GHSA-g8q2-p36q-94f6	2 months ago
George Joseph	00ebdfbc89	ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds. ... Several bounds checks have been edded to ooQ931Decode to prevent it from running past the end of the data buffer when parsing information elements. Resolves: #GHSA-746q-794h-cc7f	2 months ago
George Joseph	92a5dd1003	ARI: Make ARI applications respect live_dangerously. ... DeveloperNote: ARI applications can no longer call "dangerous" dialplan functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling "live_dangerously" in asterisk.conf. Resolves: #GHSA-vrfp-mg3q-3959	2 months ago
George Joseph	eaa108c68a	res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns. ... * Add check to red_t140_to_red() to ensure that the new primary payload can't cause the rtp_red->len array items to wrap or cause an overrun of the rtp_red->t140red_data buffer. * Add check to rtp_red_buffer() to ensure that a T.140 frame to be sent can't cause rtp_red->len array items to wrap or cause an overrun of the rtp_red->buf_data buffer. Resolves: #GHSA-vfhr-r9x9-c687 Resolves: #GHSA-j2mm-57pq-jh94	2 months ago
Roberto Paleari	07aec894a1	res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser ... Add constraint checks to prevent unauthenticated users from crashing Asterisk instance by sending a crafted inbound SIP NOTIFY request with "Content-Type: application/simple-message-summary". Resolves: #GHSA-8jw3-ccr9-xrmf	2 months ago
Mike Bradeen	a1fcfdf9b9	manager: Use remote address in user error logging ... To avoid a potential null dereference use the remote address in error logging when there is no user or the user acl fails. Resolves: #GHSA-3rhj-hhw7-m6fw	2 months ago
Mike Bradeen	fff256d053	ooh323: Prevent potential buffer overflow in trace logging ... Replace a call to vsprintf with a call to ast_vasprintf to prevent a possible buffer overflow. Resolves: #GHSA-x348-j6c9-77f3	2 months ago
Pengpeng Hou	15c66022dc	app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers ... The protocol 1 unpack helpers trusted externally controlled lengths and wrote them directly into fixed-size buffers in sms_t. Clamp the address, header, and body copies to the destination array sizes so malformed messages cannot overwrite adjacent state. Resolves: #GHSA-q9fr-m7g8-6ph5	2 months ago
Milan Kyselica	e329a6803a	res_xmpp: Fix stack buffer overflow in namespace prefix handling ... The snprintf size parameter in xmpp_action_hook() is computed from the attacker-controlled namespace prefix length and is not bounded by the 256-byte stack buffer size. When a remote XMPP peer sends a stanza with a child element whose namespace prefix exceeds 249 characters, snprintf writes past the buffer boundary. Use sizeof(attr) as the snprintf size limit and %.*s precision to extract only the prefix portion of the element name, preserving the original truncation behavior for valid inputs. Resolves: #GHSA-mxgm-8c6f-5p8f	2 months ago
Milan Kyselica	0e89514084	res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser ... The parse_simple_message_summary() function uses sscanf with an unbounded %s format specifier to parse the Message-Account field from incoming SIP NOTIFY bodies into a fixed-size 512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single unauthenticated SIP NOTIFY with a Message-Account value exceeding 512 bytes overflows the buffer, corrupting adjacent stack data and permanently disabling the PJSIP transport layer without crashing the process. Add a width specifier (%511s) to limit the sscanf write to PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator, matching the destination buffer size. Resolves: #GHSA-589g-qgf8-m6mx	2 months ago
Milan Kyselica	14968f2f44	res_config_ldap: Escape LDAP filter values per RFC 4515 ... The LDAP realtime driver constructs search filters by directly concatenating user-supplied values without RFC 4515 escaping. When LDAP is used as a realtime backend for endpoint identification, characters with special meaning in LDAP filters (*, (, ), \) can be injected via the SIP From header username. Add ldap_filter_escape_value() that escapes RFC 4515 special characters to their \HH hex representation, and apply it to non-LIKE query values. The LIKE query path preserves the existing wildcard conversion behavior with a note for maintainers. Resolves: #GHSA-r6c2-hwc2-j4mp	2 months ago
Milan Kyselica	0a86e45e8c	cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection ... The eventtype column handler in cel_pgsql.c inserts record.user_defined_name directly into the SQL query without calling PQescapeStringConn(), while all other string fields in the same function are properly escaped. Similarly, cel_tds.c passes the raw user_defined_name into the SQL INSERT without routing it through anti_injection(), while all other fields are processed through that function. For cel_pgsql.c, escape the eventtype value using PQescapeStringConn(), matching the existing pattern used for all other string fields at lines 308-331 of the same function. For cel_tds.c, route the eventtype value through anti_injection() consistent with how all other fields are handled in the same function. Resolves: #GHSA-ph27-3m5q-mj5m	2 months ago
Milan Kyselica	87399593cd	http: Escape error page text to prevent reflected XSS ... The text parameter in ast_http_create_response() is inserted into the HTML body without escaping, while the server name on the same page is properly escaped via ast_xml_escape(). When res_phoneprov passes the decoded request URI as the text of a 404 response, HTML metacharacters in the URI are rendered by the browser. Apply ast_xml_escape() to the text parameter before inserting it into the HTML template, using the same function already used for the server name. Resolves: #GHSA-4pgv-j3mr-3rcp	2 months ago
Milan Kyselica	aa40bc599a	codec_codec2: Only process complete Codec2 frames in decoder ... The codec2_samples() function uses floor division (160 * datalen/6) to compute expected output samples, but the decode loop condition (x < datalen) iterates with ceiling behavior when datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch causes the loop to decode one extra frame beyond what the framework bounds check budgeted for, leading to an out-of-bounds write on the output buffer. Change the loop condition to only process complete frames, matching the floor-division behavior of codec2_samples(). This also prevents an out-of-bounds read on the input side when fewer than CODEC2_FRAME_LEN bytes remain. Resolves: #GHSA-qf8j-jp7h-c5hx	2 months ago
Milan Kyselica	ccec146953	format_ogg_speex: Add bounds check to prevent heap buffer overflow ... The ogg_speex_read() function copies OGG packet data via memcpy() without validating the packet size against the destination buffer (BUF_SIZE = 200 bytes). A crafted .spx file with an oversized OGG audio packet causes a heap buffer overflow that corrupts the adjacent speex_desc structure containing libogg heap pointers, leading to a crash (SIGSEGV) on playback. Add a bounds check for both negative and oversized values before the memcpy, consistent with how format_ogg_vorbis bounds its reads via ov_read(). Resolves: #GHSA-8jhw-m2hg-vp3h	2 months ago
Asterisk Development Team	eb9e5f4093	Update for 23.4.0	3 months ago
Asterisk Development Team	0869f09aef	Update for 23.4.0-rc1	3 months ago
George Joseph	7c8d255b6e	res_ari: Add res_ari_model as an optional_module. ... Under certain timing/load conditions, res_ari_model may not load until after res_ari on startup or it might unload before res_ari on shutdown. This can cause a segfault when DEVMODE is enabled and there are persistent outbound websocket connections because DEVMODE forces validation of outgoing events against the models. To prevent this, res_ari_model has been added as an "optional_module" to res_ari's NODULE_INFO. This will enforce load/unload order but not make res_ari dependent on res_ari_model. However, if Asterisk is configured with --enable-dev-mode, res_ari will fail to load if res_ari_model isn't available. Resolves: #1970	3 months ago
Mike Bradeen	65a66732fb	res ari: Add attachable states to Channels and Bridges ... Adds the ability to attach multiple states to both Channels and Bridges in the form of variables that are included in all events on the associated object. First, this adds an optional boolean field to channel variables 'report_events' that causes the variable to automatically be included in all events on that channel. To allow this, variables can now be either name value pairs (the current format): `<variable_name>: '<value_string>'` - or - `<variable_name>: {value: '<value_string>', report_events: [true|false]}` If the old format is used or 'report_events' is not included, it will default to false and retain current behavior. Second, this extends both reported and unreported variables to Bridges so they too may have stateful information. Resolves: #1910 UserNote: Bridge variables now can be set and retrieved via the following paths: `/bridges/{bridgeId}/variable` `/bridges/{bridgeId}/variables` Both Bridge and Channel variables can now be set with an optional 'report_events' boolean flag that will cause those variables to be included on all events on that object. The 'report_events' flag will default to False if not set to maintain backwards capability. To allow this, variables can now be either name value pairs (the current format): `<variable_name>: '<value_string>'` - or - `<variable_name>: {value: '<value_string>', report_events: [true|false]}`	3 months ago
Ben Ford	5993291794	ARI: Added paths to get and set multiple channel variables. ... Two new paths exist for ARI to get and set multiple channel variables at the same time. This is done via GET and POST like the single get and set variable equivalents. Leading and trailing whitespace will be stripped from the variable names for both paths. When setting variables, the values will be read as-is, whitespace included. GET takes in a single string with comma-separated values, while POST takes in a dictionary of key value pairs. The code follows the same paths as when setting multiple variables when originating a channel via ARI. UserNote: Added new ARI paths for getting and setting multiple channel variables at a time. For GET, this takes in a single string of comma-separated variable names, while POST takes in a dictionary of key value pairs. The behavior is the same as passing in variables when originating a channel.	3 months ago
Bernd Kuhls	32d32d2267	res_stir_shaken: avoid direct ASN1_STRING accesses ... https://github.com/openssl/openssl/issues/29117 Signed-off-by: Bernd Kuhls <bernd@kuhls.net> Resolves: #1952	3 months ago
Bernd Kuhls	8f8afb6d1f	tcptls.c: fix build with OpenSSL 4 ... tcptls.c: In function '__ssl_setup': tcptls.c:417:52: error: implicit declaration of function 'SSLv3_client_method'; did you mean 'SSLv23_client_method'? [-Wimplicit-function-declaration] 417 | cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method()); SSLv3_client_method was removed from OpenSSL 4.0.0: https://github.com/openssl/openssl/blob/openssl-4.0.0/doc/man7/ossl-removed-api.pod?plain=1#L440 Signed-off-by: Bernd Kuhls <bernd@kuhls.net> Resolves: #1952	3 months ago
mikhail_grishak	d9ec9d7dd0	res_calendar: Fix build with libical 4.X ... libical 4.0 removed the icaltime_add() function in favor of icaltime_adjust(). Additionally, the callback signature for icalcomponent_foreach_recurrence() was updated to use a const pointer for the icaltime_span argument. This commit adds conditional compilation using ICAL_MAJOR_VERSION to support both libical 3.X and the new 4.X API, ensuring backward compatibility. Fixes: #1957	3 months ago
UpBeta	0bf5aa9e9d	app_record: Fix hangup handling during beep playback ... When a hangup occurs while app_record is playing the initial beep, the application does not detect the hangup and continues running until the maxduration timeout expires. Replace the manual ast_streamfile() + ast_waitstream() sequence with ast_stream_and_wait(), which properly detects hangup and returns non-zero, allowing the application to exit immediately with RECORD_STATUS set to HANGUP. Resolves: #1950	3 months ago
smtcbn	05f2430c02	odbc: Don't use prepared statements for distinct SQL statements ... Avoids unnecessary prepare for simple INSERT statements that cause issues with ProxySQL (prepared statement counter overflow). Resolves: #1217	3 months ago
Alexander Bakker	b25df18d32	abstract_jb.c: Remove timerfd from channel when disabling jitter buffer ... Previously, the lingering timerfd would cause a tight loop if the channel enters a BridgeWait after the jitter buffer was disabled. Fixes: #1762	3 months ago
Sean Bright	c051617685	res_pjsip: Don't allow a leading period when wildcard matching ... The reference identifier (what the client provides - in this case a hostname) must start with a domain label, not a `.`. The current implementation will match `.seanbright.com` against `*.seanbright.com` which is incorrect.	3 months ago
George Joseph	5044136026	Ensure channel locks aren't held while calling ast_set_variables. ... If the channel is locked when calling ast_set_variables and any of the variables contained dialplan functions, there's a possiblilty of a deadlock. To prevent this, either the explicit locks were removed or the call to ast_set_variables moved out of the lock scope. A warning to not hold channel locks is also added to the documentation for ast_set_variables. Resolves: #1936	3 months ago
smtcbn	c85b0d0f48	app_queue: fix double increment of member->calls with shared_lastcall ... Under high concurrency, update_queue() may be invoked multiple times for the same call, causing member->calls and queue-level counters to be incremented more than once. The existing starttime check is not atomic and allows concurrent execution paths to pass. Treat member->starttime as a single-use token and consume it via CAS to ensure the call is counted exactly once. This also prevents incorrect call distribution when using strategies such as fewestcalls. Observed as a regression after upgrading to 20.17. Resolves: #1736	3 months ago
George Joseph	3a0ab15e91	chan_dahdi: Fix set but not used in mfcr2_show_links_of(). ... When openr2 is installed mfcr2_show_links_of() is no longer ifdeffed out which makes gcc-16 complain with 'variable ‘x’ set but not used'. Resolves: #1947	3 months ago
Sebastian Jennen	19fb0a972f	tests: add tests/test_codec_translations.c ... This tests checks [slin -> codec -> slin] and then compares slin in vs out regarding signal noise ratio and delay. Near-lossless codecs (ulaw, alaw) are checked with a maximum per-sample error bound. Lossy codecs are checked with a per-codec SNR threshold. Cross-correlation alignment compensates for algorithmic delay in codecs like speex and opus. Covered codecs: ulaw, alaw, adpcm, g726, g726aal2, gsm, speex, speex16, speex32, ilbc, codec2, lpc10, g722, opus. Resolves: #1812	3 months ago
Sean Bright	1db7cad81d	install_prereq: Add a 'minimal' mode for basic build dependencies	3 months ago
George Joseph	19bc22d480	chan_websocket: Handle incoming CONTINUATION frames. ... chan_websocket now tells res_http_websocket to accumulate incoming CONTINUATION frames into 1024 byte TEXT or BINARY frames. Resolves: #1941	3 months ago
George Joseph	24ba87d684	res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). ... ``` AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ rtp->rtcp->stdev_rxjitter); ``` Should have been ``` AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ rtp->rtcp->stdev_rxmes); ``` Note the last macro parameter name. Resolves: #1938	3 months ago
Stanislav Abramenkov	33e043c1c5	jansson: Upgrade version to jansson 2.15.0 ... UpgradeNote: jansson has been upgraded to 2.15.0. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0 Resolves: #1931	3 months ago
George Joseph	909420a556	channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. ... The original trigger for setting the RTP stats in ast_softhangup() came from an ARI issue where stats weren't being set in time to be reported on STASIS_END events. The thought was that setting them in a common place like ast_softhangup() would ensure the stats were set in possibly other scenarios. Unfortunately, setting the RTP stats variables in ast_softhangup() broke ABI as it required that no channel locks be held which was not the case earlier. Given that the original issue was ARI, we can move setting the stats to ast_ari_channels_hangup() in resource_channels just before it calls ast_softhangup(). This might not catch all cases of the stats not being set, but it won't break ABI or deadlock either. Resolves: #1928	3 months ago
George Joseph	1f064911ed	res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 ... If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution is enabled based on the TTL returned in the DNS results. If the TTL returned is 0, it means that the next time the IP address is needed, it must be looked up again. I.E. Don't cache. Historically (and incorrectly) however, res_rtp_asterisk stopped the periodic resolution and never re-resolved the hostname again. Besides what's mentioned in the user notes... * Additional debugging was added in various STUN/DNS functions. * The `rtp show settings` CLI command shows more detailed STUN info. * Some debugging was added to dns_core.c and dns_recurring.c. UserNote: A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf that allows control over what happens when a STUN server hostname lookup returns a TTL of 0. The values can be set as follows: - 'no': This is the historical (and current default) behavior of not doing any further lookups and continuing to use the last successful result until Asterisk is restarted or rtp.conf is reloaded. - 'yes': Use the last cached result for the current call but trigger re-resolution in the background for the benefit of future calls. If the result of the background lookup is a ttl > 0, periodic resolution will be restarted otherwise the next call will use the new cached value and will trigger a background lookup again. UserNote: A new CLI command `rtp resolve stun hostname` has been added that will force a resolution of the STUN hostname and (re)start periodic resolution if the result has a TTL > 0. Resolves: #1858	3 months ago
Jaco Kroon	c943b0e2a1	pjsip_configuration: Show actual dtls_verify config. ... Rather than merely showing dtls_verify : Yes/No in pjsip show endpoint xxx it will now be shown what exactly is being checked, ie, one of: dtls_verify : No dtls_verify : Fingerprint dtls_verify : Certificate dtls_verify : Yes Where Yes implies both Fingerprint and Certificate. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	3 months ago
Naveen Albert	791713d2c4	app_dial: Properly handle callee hangup while sending digits. ... If we are sending digits (either DTMF, MF, or SF) to the called channel after receiving progress or a wink, and the callee hangs up before we have finished sending it digits, there are several problems that can ensue: * If the callee hung up without answering, the calling channel would hang up and not continue in the dialplan. * If the callee *did* answer before hanging up, the answer was never passed through to the caller, since this gets "eaten" by the various digit streaming functions and is never processed by app_dial. This is generally an edge case that occurs due to some kind of signaling failure, but to better handle this: * Set to_answer to 0 to prevent hangup on the exit path, just like other parts of wait_for_answer. * Better document this usage of to_answer. * If the channel did answer while it was receiving digits, manually answer the calling channel before we abort. The call would not continue in the dialplan anyways (either before or after this fix), but technically the call was answered, so the CDRs should probably reflect that, and this mirrors the behavior of calls which normally do not continue. Resolves: #1915 UserNote: If a called channel sends progress or wink and the caller begins sending digits but the callee answers and then hangs up before digit sending can finish, the call is now answered before being disconnected. If the callee hangs up without answering, the call now continues in the dialplan.	3 months ago
Maximilian Fridrich	37ec65b7d3	res_pjsip_messaging: Update To URI only if it is a SIP(S) URI ... When a message is sent via ARI, the ARI endpoint only provides a To field which is also used as destination field. This means that the To field might not necessarily contain a SIP URI but might instead specify an Asterisk endpoint (in MessageDestinationInfo format). This led to many warnings even though the message was sent correctly. The fix is to only call `ast_sip_update_to_uri` if the To field starts with the sip: or sips: scheme. Resolves: #1357	3 months ago
Stanislav Abramenkov	4c05cd0bfa	Upgrade bundled pjproject to 2.17. ... Resolves: #1888 UserNote: Bundled pjproject has been upgraded to 2.17. For more information about what is included in this release, see the pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17	3 months ago