4339 Commits (a7093ba51f7a324feaebb36342eded5ded59e90c)

Author	SHA1	Message	Date
George Joseph	fded9fc8a4	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	7fdff94861	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 (cherry picked from commit 23c4d21d1b)	2 years ago
Sean Bright	e6e0ec9105	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 * AC_HEADER_TIME is obsolete.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 (cherry picked from commit a36af23f20)	2 years ago
George Joseph	a9da65d838	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 (cherry picked from commit acb18c1fc4)	2 years ago
Naveen Albert	c3b04a314c	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 (cherry picked from commit 8a03ed6877)	2 years ago
Maximilian Fridrich	8b81c5a16b	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 (cherry picked from commit a4cd452246)	2 years ago
Sean Bright	0e92662c29	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 (cherry picked from commit 3e2a28fc3d)	2 years ago
Mike Bradeen	1cba481518	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. (cherry picked from commit fa18f2d71e)	2 years ago
Sean Bright	a7f1dbc462	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 (cherry picked from commit 6f218514fb)	2 years ago
Maximilian Fridrich	7140faf410	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 (cherry picked from commit ecd5f91125)	2 years ago
Naveen Albert	68a1c9b313	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 (cherry picked from commit 4e602a1afe)	2 years ago
Naveen Albert	77384b93a1	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 (cherry picked from commit 1bbcb98558)	2 years ago
Mike Bradeen	03da795e3e	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 (cherry picked from commit b2e9419961)	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	3 years ago
Mike Bradeen	24e27a3c6e	res_pjsip: Upgraded bundled pjsip to 2.13 ... Removed multiple patches. Code chages in res_pjsip_pubsub due to changes in evsub. Pjsip now calls on_evsub_state() before on_rx_refresh(), so the sub tree deletion that used to take place in on_evsub_state() now must take place in on_rx_refresh(). Additionally, pjsip now requires that you send the NOTIFY from within on_rx_refresh(), otherwise it will assert when going to send the 200 OK. The idea is that it will look for this NOTIFY and cache it until after sending the response in order to deal with the self-imposed message mis-order. Asterisk previously dealt with this by pushing the NOTIFY in on_rx_refresh(), but pjsip now forces us to use it's method. Changes were required to configure in order to detect which way pjsip handles this as the two are not compatible for the reasons mentioned above. A corresponding change in testsuite is required in order to deal with the small interal timing changes caused by moving the NOTIFY send. ASTERISK-30325 Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f	3 years ago
Sean Bright	41d3a57627	doxygen: Fix doxygen errors. ... Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
Naveen Albert	ec0ca7dcbc	res_pjsip_session: Add overlap_context option. ... Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	3 years ago
Igor Goncharovsky	de745157ca	res_pjsip_rfc3326: Add SIP causes support for RFC3326 ... Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	3 years ago
George Joseph	345ff2d8ee	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	3 years ago
George Joseph	8067229418	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit 62745013a4. Reason for revert: Issue when transcoding to/from g722 Change-Id: I1665a5442bfb6d7bfa06fdcea3374f4581395b4a	3 years ago
Naveen Albert	811584ded3	json.h: Add ast_json_object_real_get. ... json.h contains macros to get a string and an integer from a JSON object. However, the macro to do this for JSON reals is missing. This adds that. ASTERISK-30361 #close Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a	3 years ago
George Joseph	62745013a4	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	3 years ago
Naveen Albert	40ed776c40	res_pjsip_session: Use Caller ID for extension matching. ... Currently, there is no Caller ID available to us when checking for an extension match when handling INVITEs. As a result, extension patterns that depend on the Caller ID are not matched and calls may be incorrectly rejected. The Caller ID is not available because the supplement that adds Caller ID to the session does not execute until after this check. Supplement callbacks cannot yet be executed at this point since the session is not yet in the appropriate state. To fix this without impacting existing behavior, the Caller ID number is now retrieved before attempting to pattern match. This ensures pattern matching works correctly and there is no behavior change to the way supplements are called. ASTERISK-28767 #close Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7	3 years ago
Peter Fern	b1f1556922	streams: Ensure that stream is closed in ast_stream_and_wait on error ... When ast_stream_and_wait returns an error (for example, when attempting to stream to a channel after hangup) the stream is not closed, and callers typically do not check the return code. This results in leaking file descriptors, leading to resource exhaustion. This change ensures that the stream is closed in case of error. ASTERISK-30198 #close Reported-by: Julien Alie Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803	3 years ago
Michael Kuron	d73f22268f	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip ... chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333	3 years ago
Naveen Albert	0fb48fb751	rtp_engine.h: Update examples using ast_format_set. ... This file includes some doxygen comments referencing ast_format_set. This is an obsolete API that was removed years back, but documentation was not fully updated to reflect that. These examples are updated to the current way of doing things (using the format cache). ASTERISK-30327 #close Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7	3 years ago
George Joseph	5741569c5e	pjsip_transport_events: Fix possible use after free on transport ... It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	3 years ago
Mike Bradeen	76d10ea668	manager: prevent file access outside of config dir ... Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404	3 years ago
Henning Westerholt	093593a7c8	res_pjsip: return all codecs on a re-INVITE without SDP ... Currently chan_pjsip on receiving a re-INVITE without SDP will only return the codecs that are previously negotiated and not offering all enabled codecs. This causes interoperability issues with different equipment (e.g. from Cisco) for some of our customers and probably also in other scenarios involving 3PCC infrastructure. According to RFC 3261, section 14.2 we SHOULD return all codecs on a re-INVITE without SDP The PR proposes a new parameter to configure this behaviour: all_codecs_on_empty_reinvite. It includes the code, documentation, alembic migrations, CHANGES file and example configuration additions. ASTERISK-30193 #close Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148	3 years ago
Mike Bradeen	4fd2f07ff9	audiohook: add directional awareness ... Add enum to allow setting optional direction. If set to only one direction, only feed matching-direction frames to the associated slin factory. This prevents mangling the transcoder on non-mixed frames when the READ and WRITE frames would have otherwise required it. Also removes the need to mute or discard the un-wanted frames as they are no longer added in the first place. res_stasis_snoop is changed to use this addition to set direction on audiohook based on spy direction. If no direction is set, the ast_audiohook_init will init this enum to BOTH which maintains existing functionality. ASTERISK-30252 Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb	3 years ago
Naveen Albert	faca1fcfd3	cdr: Allow bridging and dial state changes to be ignored. ... Allows bridging, parking, and dial messages to be globally ignored for all CDRs such that only a single CDR record is generated per channel. This is useful when CDRs should endure for the lifetime of an entire channel and bridging and dial updates in the dialplan should not result in multiple CDR records being created for the call. With the ignore bridging option, bridging changes have no impact on the channel's CDRs. With the ignore dial state option, multiple Dials and their outcomes have no impact on the channel's CDRs. The last disposition on the channel is preserved in the CDR, so the actual disposition of the call remains available. These two options can reduce the amount of "CDR hacks" that have hitherto been necessary to ensure that CDR was not "spoiled" by these messages if that was undesired, such as putting a dummy optimization-disabled local channel between the caller and the actual call and putting the CDR on the channel in the middle to ensure that CDR would persist for the entire call and properly record start, answer, and end times. Enabling these options is desirable when calls correspond to the entire lifetime of channels and the CDR should reflect that. Current default behavior remains unchanged. ASTERISK-30091 #close Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832	3 years ago
Maximilian Fridrich	675bed5f2c	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	3 years ago
George Joseph	96bf50b146	manager.h: Bump version to 7.0.3 ... Change-Id: I5c365e141769f8ce2a1aeb4ae2cdb9842dd2fd5b	3 years ago
Philip Prindeville	ddc8f6967f	test: initialize capture structure before freeing ... ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	3 years ago
Maximilian Fridrich	479ce16eba	res_pjsip: Add 100rel option "peer_supported". ... This patch adds a new option to the 100rel parameter for pjsip endpoints called "peer_supported". When an endpoint with this option receives an incoming request and the request indicated support for the 100rel extension, then Asterisk will send 1xx responses reliably. If the request did not indicate 100rel support, Asterisk sends 1xx responses normally. ASTERISK-30158 Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad	3 years ago
George Joseph	4ab3cd6f86	res_geolocation: Fix segfault when there's an empty element ... Fixed a segfault caused by var_list_from_loc_info() encountering an empty location info element. Fixed an issue in ast_strsep() where a value with only whitespace wasn't being preserved. Fixed an issue in ast_variable_list_from_quoted_string() where an empty value was considered a failure. ASTERISK-30215 Reported by: Dan Cropp Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a	3 years ago
Ben Ford	1f685d6969	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	3 years ago
Philip Prindeville	bb8b2259bc	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	3 years ago
Philip Prindeville	0972106d51	test: Add coverage for res_crypto ... We're validating the following functionality: encrypting a block of data with RSA decrypting a block of data with RSA signing a block of data with RSA verifying a signature with RSA encrypting a block of data with AES-ECB encrypting a block of data with AES-ECB as well as accessing test keys from the keystore. ASTERISK-30045 #close Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d	3 years ago
Philip Prindeville	53f8cc1d49	res_crypto: make keys reloadable on demand for testing ... ASTERISK-30045 Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8	3 years ago
Philip Prindeville	aeb23fe2a7	main/utils: allow checking for command in $PATH ... ASTERISK-30037 Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf	3 years ago
Philip Prindeville	f2cd48e9a8	test: Add ability to capture child process output ... ASTERISK-30037 Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2	3 years ago
George Joseph	0e9076df7b	res_geolocation: Add two new options to GEOLOC_PROFILE ... Added an 'a' option to the GEOLOC_PROFILE function to allow variable lists like location_info_refinement to be appended to instead of replacing the entire list. Added an 'r' option to the GEOLOC_PROFILE function to resolve all variables before a read operation and after a Set operation. Added a few missing parameters to the ones allowed for writing with GEOLOC_PROFILE. Fixed a bug where calling GEOLOC_PROFILE to read a parameter might actually update the profile object. Cleaned up XML documentation a bit. ASTERISK-30190 Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c	3 years ago
George Joseph	261051ecec	res_geolocation: Allow location parameters on the profile object ... You can now specify the location object's format, location_info, method, location_source and confidence parameters directly on a profile object for simple scenarios where the location information isn't common with any other profiles. This is mutually exclusive with setting location_reference on the profile. Updated appdocsxml.dtd to allow xi:include in a configObject element. This makes it easier to link to complete configOptions in another object. This is used to add the above fields to the profile object without having to maintain the option descriptions in two places. ASTERISK-30185 Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569	3 years ago
George Joseph	372f5c56e1	res_geolocation: Add profile parameter suppress_empty_ca_elements ... Added profile parameter "suppress_empty_ca_elements" that will cause Civic Address elements that are empty to be suppressed from the outgoing PIDF-LO document. Fixed a possible SEGV if a sub-parameter value didn't have a value. ASTERISK-30177 Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd	3 years ago
Naveen Albert	8041bd12c1	cli: Prevent assertions on startup from bad ao2 refs. ... If "core show channels" is run before startup has completed, it is possible for bad ao2 refs to occur because the system is not yet fully initialized. This will lead to an assertion failing. To prevent this, initialization of CLI builtins is moved to be later along in the main load sequence. Core CLI commands are loaded at the same time, but channel-related commands are loaded later on. ASTERISK-29846 #close Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6	3 years ago
Joshua C. Colp	f97e9aed85	pjsip: Add TLS transport reload support for certificate and key. ... This change adds support using the pjsip_tls_transport_restart function for reloading the TLS certificate and key, if the filenames remain unchanged. This is useful for Let's Encrypt and other situations. Note that no restart of the transport will occur if the certificate and key remain unchanged. ASTERISK-30186 Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0	3 years ago
Sean Bright	57972bac38	channel.h: Remove redundant declaration. ... The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set() for us, so no need to declare it separately. Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a	3 years ago
Naveen Albert	6000f9963c	features: Add transfer initiation options. ... Adds additional control options over the transfer feature functionality to give users more control in how the transfer feature sounds and works. First, the "transfer" sound that plays when a transfer is initiated can now be customized by the user in features.conf, just as with the other transfer sounds. Secondly, the user can now specify the transfer extension in advance by using the TRANSFER_EXTEN variable. If a valid extension is contained in this variable, the call will automatically be transferred to this destination. Otherwise, it will fall back to collecting the extension from the user as is always done now. ASTERISK-29899 #close Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3	3 years ago
Naveen Albert	9126f9bd2b	general: Very minor coding guideline fixes. ... Fixes a few coding guideline violations: * Use of C99 comments * Opening brace on same line as function prototype ASTERISK-30163 #close Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7	3 years ago