370 Commits (18)

Author	SHA1	Message	Date
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	2 years ago
Naveen Albert	3babc68099	res_hep: Add support for named capture agents. ... Adds support for the capture agent name field of the Homer protocol to Asterisk by allowing users to specify a name that will be sent to the HEP server. ASTERISK-30322 #close Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b	2 years ago
Naveen Albert	5a78ce5a1f	app_voicemail: Fix missing email in msg_create_from_file. ... msg_create_from_file currently does not dispatch emails, which means that applications using this function, such as MixMonitor, will not trigger notifications to users (only AMI events are sent our currently). This is inconsistent with other ways users can receive voicemail. This is fixed by adding an option that attempts to send an email and falling back to just the notifications as done now if that fails. The existing behavior remains the default. ASTERISK-30283 #close Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7	2 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	2 years ago
Naveen Albert	0bbcda3040	chan_dahdi: Allow FXO channels to start immediately. ... Currently, chan_dahdi will wait for at least one ring before an incoming call can enter the dialplan. This is generally necessary in order to receive the Caller ID spill and/or distinctive ringing detection. However, if neither of these is required, then there is nothing gained by waiting for one ring and this unnecessarily delays call setup. Users can now use immediate=yes to make FXO channels (FXS signaled) begin processing dialplan as soon as Asterisk receives the call. ASTERISK-30305 #close Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5	2 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	2 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	2 years ago
George Joseph	7e79d93ed3	res_geolocation: Update wiki documentation ... Also added a note to the geolocation.conf.sample file and added a README to the res/res_geolocation/wiki directory. Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc	2 years ago
Naveen Albert	fdac31f279	app_amd: Add option to play audio during AMD. ... Adds an option that will play an audio file to the party while AMD is running on the channel, so the called party does not just hear silence. ASTERISK-30179 #close Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6	2 years ago
sungtae kim	2f5cfa548f	res_musiconhold: Add option to not play music on hold on unanswered channels ... This change adds an option, answeredonly, that will prevent music on hold on channels that are not answered. ASTERISK-30135 Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08	2 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	2 years ago
Naveen Albert	8c6c06a340	app_confbridge: Add end_marked_any option. ... Adds the end_marked_any option, which can be used to kick a user from a conference if any marked user leaves. ASTERISK-30211 #close Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5	2 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	2 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	2 years ago
George Joseph	1fc0298f87	res_geolocation: Add built-in profiles ... The trigger to perform outgoing geolocation processing is the presence of a geoloc_outgoing_call_profile on an endpoint. This is intentional so as to not leak location information to destinations that shouldn't receive it. In a totally dynamic configuration scenario however, there may not be any profiles defined in geolocation.conf. This makes it impossible to do outgoing processing without defining a "dummy" profile in the config file. This commit adds 4 built-in profiles: "<prefer_config>" "<discard_config>" "<prefer_incoming>" "<discard_incoming>" The profiles are empty except for having their precedence set and can be set on an endpoint to allow processing without entries in geolocation.conf. "<discard_config>" is actually the best one to use in this situation. ASTERISK-30182 Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a	2 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	2 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	2 years ago
George Joseph	dfedb13eca	res_geolocation: Address user issues, remove complexity, plug leaks ... * Added processing for the 'confidence' element. * Added documentation to some APIs. * removed a lot of complex code related to the very-off-nominal case of needing to process multiple location info sources. * Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes one eprofile instead of a datastore of multiples. * Plugged a huge leak in XML processing that arose from insufficient documentation by the libxml/libxslt authors. * Refactored stylesheets to be more efficient. * Renamed 'profile_action' to 'profile_precedence' to better reflect it's purpose. * Added the config option for 'allow_routing_use' which sets the value of the 'Geolocation-Routing' header. * Removed the GeolocProfileCreate and GeolocProfileDelete dialplan apps. * Changed the GEOLOC_PROFILE dialplan function as follows: * Removed the 'profile' argument. * Automatically create a profile if it doesn't exist. * Delete a profile if 'inheritable' is set to no. * Fixed various bugs and leaks * Updated Asterisk WiKi documentation. ASTERISK-30167 Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0	2 years ago
Naveen Albert	d0cd6e82a6	cdr.conf: Remove obsolete app_mysql reference. ... The CDR sample config still mentions that app_mysql is available in the addons directory, but this is incorrect as it was removed as of 19. This removes that to avoid confusion. ASTERISK-30160 #close Change-Id: Ie5293ccb4f2b365896981811b480544e67bb9cd7	2 years ago
Sam Banks	56c6d76a45	queues.conf.sample: Correction of typo ... ASTERISK-30126 #close Change-Id: I009c4dcbf9338a13e3baf87b52a5bbe4f9f81a42	2 years ago
George Joseph	cdbd973ecd	Geolocation: chan_pjsip Capability Preview ... This commit adds res_pjsip_geolocation which gives chan_pjsip the ability to use the core geolocation capabilities. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30128 Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9	2 years ago
George Joseph	f491355384	Geolocation: Core Capability Preview ... This commit adds res_geolocation which creates the core capabilities to manipulate Geolocation information on SIP INVITEs. An upcoming commit will add res_pjsip_geolocation which will allow the capabilities to be used with the pjsip channel driver. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30127 Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303	2 years ago
Naveen Albert	0fe1a581e5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	2 years ago
Kevin Harwell	0ddbf6bc45	res_pjsip: allow TLS verification of wildcard cert-bearing servers ... Rightly the use of wildcards in certificates is disallowed in accordance with RFC5922. However, RFC2818 does make some allowances with regards to their use when using subject alt names with DNS name types. As such this patch creates a new setting for TLS transports called 'allow_wildcard_certs', which when it and 'verify_server' are both enabled allows DNS name types, as well as the common name that start with '*.' to match as a wildcard. For instance: *.example.com will match for: foo.example.com Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc... And the starting wildcard only matches for a single level. For instance: *.example.com will NOT match for: foo.bar.example.com The new setting is disabled by default. ASTERISK-30072 #close Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4	2 years ago
Naveen Albert	b81fbbc6dc	res_pjsip_outbound_registration: Make max random delay configurable. ... Currently, PJSIP will randomly wait up to 10 seconds for each outbound registration's initial attempt. The reason for this is to avoid having all outbound registrations attempt to register simultaneously. This can create limitations with the test suite where we need to be able to receive inbound calls potentially within 10 seconds of starting up. For instance, we might register to another server and then try to receive a call through the registration, but if the registration hasn't happened yet, this will fail, and hence this inconsistent behavior can cause tests to fail. Ultimately, this requires a smaller random value because there may be no good reason to wait for up to 10 seconds in these circumstances. To address this, a new config option is introduced which makes this maximum delay configurable. This allows, for instance, this to be set to a very small value in test systems to ensure that registrations happen immediately without an unnecessary delay, and can be used more generally to control how "tight" the initial outbound registrations are. ASTERISK-29965 #close Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd	2 years ago
Michael Cargile	216a55408e	apps/confbridge: Added hear_own_join_sound option to control who hears sound_join ... Added the hear_own_join_sound option to the confbridge user profile to control who hears the sound_join audio file. When set to 'yes' the user entering the conference and the participants already in the conference will hear the sound_join audio file. When set to 'no' the user entering the conference will not hear the sound_join audio file, but the participants already in the conference will hear the sound_join audio file. ASTERISK-29931 Added by Michael Cargile Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2	2 years ago
Naveen Albert	cb53ad5671	chan_dahdi: Don't append cadences on dahdi restart. ... Currently, if any custom ring cadences are specified, they are appended to the array of cadences from wherever we left off last time. This works properly the first time, but on subsequent dahdi restarts, it means that the existing cadences are left alone and (most likely) the same cadences are then re-added afterwards. In short order, the cadence array gets maxed out and the user begins seeing warnings that the array is full and no more cadences may be added. This buggy behavior persists until Asterisk is completely restarted; however, if and when dahdi restart is run again, then the same problem is reintroduced. This fixes this behavior so that cadence parsing is more idempotent, that is so running dahdi restart multiple times starts adding cadences from the beginning, rather than from wherever the last cadence was added. As before, it is still not possible to revert to the default cadences by simply removing all cadences in this manner, nor is it possible to delete existing cadences. However, this does make it possible to update existing cadences, which was not possible before, and also ensures that the cadences remain unchanged if the config remains unchanged. ASTERISK-29990 #close Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d	2 years ago
Mark Petersen	16e59db514	chan_pjsip: add allow_sending_180_after_183 option ... added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	2 years ago
Kevin Harwell	2fb8667908	res_aeap & res_speech_aeap: Add Asterisk External Application Protocol ... Add framework to connect to, and read and write protocol based messages from and to an external application using an Asterisk External Application Protocol (AEAP). This has been divided into several abstractions: 1. transport - base communication layer (currently websocket only) 2. message - AEAP description and data (currently JSON only) 3. transaction - links/binds requests and responses 4. aeap - transport, message, and transaction handler/manager This patch also adds an AEAP implementation for speech to text. Existing speech API callbacks for speech to text have been completed making it possible for Asterisk to connect to a configured external translator service and provide audio for STT. Results can also be received from the external translator, and made available as speech results in Asterisk. Unit tests have also been created that test the AEAP framework, and also the speech to text implementation. ASTERISK-29726 #close Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2	2 years ago
Ben Ford	62f8e157fb	res_aeap: Add basic config skeleton and CLI commands. ... Added support for a basic AEAP configuration read from aeap.conf. Also added 2 CLI commands for showing individual configurations as well as all of them: aeap show server <id> and aeap show servers. Only one configuration option is required at the moment, and that one is server_url. It must be a websocket URL. The other option, codecs, is optional and will be used over the codecs specified on the endpoint if provided. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453 Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3 (cherry picked from commit 45a1977de4)	2 years ago
Joshua C. Colp	09e8667fa5	res_pjsip: Always set async_operations to 1. ... The async_operations setting on a transport configures how many simultaneous incoming packets the transport can handle when multiple threads are polling and waiting on the transport. As we only use a single thread this was needlessly creating incoming packets when set to a non-default value, wasting memory. ASTERISK-30006 Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f	2 years ago
Ben Ford	11accf8064	AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. ... Adds a new configuration option, stir_shaken_profile, in pjsip.conf that can be specified on a per endpoint basis. This option will reference a stir_shaken_profile that can be configured in stir_shaken.conf. The type of this option must be 'profile'. The stir_shaken option can be specified on this object with the same values as before (attest, verify, on), but it cannot be off since having the profile itself implies wanting STIR/SHAKEN support. You can also specify an ACL from acl.conf (along with permit and deny lines in the object itself) that will be used to limit what interfaces Asterisk will attempt to retrieve information from when reading the Identity header. ASTERISK-29476 Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406	2 years ago
Joshua C. Colp	39cd09c246	func_odbc: Add SQL_ESC_BACKSLASHES dialplan function. ... Some databases depending on their configuration using backslashes for escaping. When combined with the use of ' this can result in a broken func_odbc query. This change adds a SQL_ESC_BACKSLASHES dialplan function which can be used to escape the backslashes. This is done as a dialplan function instead of being always done as some databases do not require this, and always doing it would result in incorrect data being put into the database. ASTERISK-29838 Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d	2 years ago
Naveen Albert	b7edc08e33	app_queue: Add music on hold option to Queue. ... Adds the m option to the Queue application, which allows a music on hold class to be specified at runtime which will override the class configured in queues.conf. This option functions like the m option to Dial. ASTERISK-29876 #close Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7	2 years ago
Naveen Albert	59b25e9d59	ami: Allow events to be globally disabled. ... The disabledevents setting has been added to the general section in manager.conf, which allows users to specify events that should be globally disabled and not sent to any AMI listeners. This allows for processing of these AMI events to end sooner and, for frequent AMI events such as Newexten which users may not have any need for, allows them to not be processed. Additionally, it also cleans up core debug as previously when debug was 3 or higher, the debug was constantly spammed by "Analyzing AMI event" messages along with a complete dump of the event contents (often for Newexten). ASTERISK-29853 #close Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205	3 years ago
Naveen Albert	8631e00a41	configs, LICENSE: remove pbx.digium.com. ... pbx.digium.com no longer accepts IAX2 calls and there are no plans for it to come back. Accordingly, nonworking IAX2 URIs are removed from both the LICENSE file and the sample config. ASTERISK-29923 #close Change-Id: I257c54d4d812ed6b4bd4cbec2cd7ebe2b87b5bad	3 years ago
Alexei Gradinari	e2423c6f49	res_pjsip_pubsub: provide a display name for RLS subscriptions ... Whereas BLFs allow to show a display name for each RLS entry, the asterisk provides only the extension now. This is not end user friendly. This commit adds a new resource_list option, resource_display_name, to indicate whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. ASTERISK-29891 #close Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681	3 years ago
Mark Petersen	6659e502a4	res_prometheus.c: missing module dependency ... added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO which fixes issue with module crashes on load "FRACK!, Failed assertion" ASTERISK-29871 Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096	3 years ago
Sean Bright	de0c29de55	res_pjsip.c: Correct minor typos in 'realm' documentation. ... Change-Id: I886936b808def5540d40071321e72f6bfa19063a	3 years ago
Naveen Albert	6fc8453e96	cdr: allow disabling CDR by default on new channels ... Adds a new option, defaultenabled, to the CDR core to control whether or not CDR is enabled on a newly created channel. This allows CDR to be disabled by default on new channels and require the user to explicitly enable CDR if desired. Existing behavior remains unchanged. ASTERISK-29808 #close Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1	3 years ago
Naveen Albert	54f2f1e027	say.conf: fix 12pm noon logic ... Fixes 12pm noon incorrectly returning 0/a.m. Also fixes a misspelling typo in the config. ASTERISK-29695 #close Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406	3 years ago
Naveen Albert	f133ae6ca2	ami: Add AMI event for Wink ... Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	3 years ago
Naveen Albert	c6410dc4ed	configs: Updates to sample configs ... Includes some minor updates to extensions.conf and iax.conf. In particular, the demonstration of macros in extensions.conf is removed, as Macro is deprecated and will be removed soon. These examples have been replaced with examples demonstrating the usage of Gosub instead. The older exten => ...,n syntax is also mostly replaced with the same keyword to demonstrate the newer, more concise way of defining extensions. IAXTEL no longer exists, so this example is replaced with something more generic. Some documentation is also added to extensions.conf and iax.conf to clarify some of the new expanded encryption capabilities with IAX2. ASTERISK-29758 #close Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46	3 years ago
Josh Soref	ae83d927d8	configs: Spelling fixes ... Correct typos of the following word families: password excludes undesirable checksums through screening interpreting database causes initiation member busydetect defined severely throughput recognized counter require indefinitely accounts ASTERISK-29714 Change-Id: Ie8f2a7b274a162dd627ee6a2165f5e8a3876527e	3 years ago
George Joseph	08cb67251f	ast_coredumper: Refactor to better find things ... The search for a running asterisk when --running is used has been greatly simplified and in the event it doesn't work, you can now specify a pid to use on the command line with --pid. The search for asterisk modules when --tarball-coredumps is used has been enhanced to have a better chance of finding them and in the event it doesn't work, you can now specify --libdir on the command line to indicate the library directory where they were installed. The DATEFORMAT variable was renamed to DATEOPTS and is now passed to the 'date' utility rather than running DATEFORMAT as a command. The coredump and output files are now renamed with DATEOPTS. This can be disabled by specifying --no-rename. Several confusing and conflicting options were removed: --append-coredumps --conffile --no-default-search --tarball-uniqueid The script was re-structured to make it easier for follow. Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59	3 years ago
Matthew Kern	15e432220c	res_pjsip_t38: bind UDPTL sessions like RTP ... In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the fallback use of the transport's bind address solve problems sending media on systems that cannot send ipv4 packets on ipv6 sockets, and certain other situations. This change extends both of these behaviors to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific problems on these systems, introducing a new option endpoint/t38_bind_udptl_to_media_address. ASTERISK-29402 Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557	3 years ago
Joseph Nadiv	4368764032	res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts ... The behavior of max_contacts and remove_existing are connected. If remove_existing is enabled, the soonest expiring contacts are removed. This may occur when there is an unavailable contact. Similarly, when remove_existing is not enabled, registrations from good endpoints are rejected in favor of retaining unavailable contacts. This commit adds a new AOR option remove_unavailable, and the effect of this setting will depend on remove_existing. If remove_existing is set to no, we will still remove unavailable contacts when they exceed max_contacts, if there are any. If remove_existing is set to yes, we will prioritize the removal of unavailable contacts before those that are expiring soonest. ASTERISK-29525 Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784	3 years ago
Naveen Albert	a65bb134f5	logger: Add custom logging capabilities ... Adds the ability for users to log to custom log levels by providing custom log level names in logger.conf. Also adds a logger show levels CLI command. ASTERISK-29529 Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702	3 years ago
Sebastien Duthil	ac492f2ff8	res_rtp_asterisk: Automatically refresh stunaddr from DNS ... This allows the STUN server to change its IP address without having to reload the res_rtp_asterisk module. The refresh of the name resolution occurs first when the module is loaded, then recurringly, slightly after the previous DNS answer TTL expires. ASTERISK-29508 #close Change-Id: I7955a046293f913ba121bbd82153b04439e3465f	3 years ago
Sarah Autumn	241686f860	sig_analog: Changes to improve electromechanical signalling compatibility ... This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	3 years ago