asterisk

Commit Graph

Author	SHA1	Message	Date
Nathan Bruning	f0df2eba9d	app_queue: Add force_longest_waiting_caller option. This adds an option 'force_longest_waiting_caller' which changes the global behavior of the queue engine to prevent queue callers from 'jumping ahead' when an agent is in multiple queues. Resolves: #108 Also closes old asterisk issues: - ASTERISK-17732 - ASTERISK-17570 Change-Id: I0f84e27903fefbe2018d0afa2d67b23aa0b321ce (cherry picked from commit `6084bbfe0e`)	2 years ago
Naveen Albert	8cdf711531	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. (cherry picked from commit `edf488c76e`)	2 years ago
Naveen Albert	f40e50ce19	sig_analog: Add fuller Caller ID support. A previous change, ASTERISK_29991, made it possible to send additional Caller ID parameters that were not previously supported. This change adds support for analog DAHDI channels to now be able to receive these parameters for on-hook Caller ID, in order to enhance the usability of CPE that support these parameters. Resolves: #94 ASTERISK-30331 UserNote: Additional Caller ID properties are now supported on incoming calls to FXS stations, namely the redirecting reason and call qualifier. (cherry picked from commit `87f44dc0f9`)	2 years ago
Sean Bright	7677e78b4f	core: Cleanup gerrit and JIRA references. (#57 ) * 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 `5c6d5ea38f`)	3 years ago
Naveen Albert	e6b84eca75	chan_dahdi: Add dialmode option for FXS lines. Currently, both pulse and tone dialing are always enabled on all FXS lines, with no way of disabling one or the other. In some circumstances, it is desirable or necessary to disable one of these, and this behavior can be problematic. A new "dialmode" option is added which allows setting the methods to support on a per channel basis for FXS (FXO signalled lines). The four options are "both", "pulse", "dtmf"/"tone", and "none". Additionally, integration with the CHANNEL function is added so that this setting can be updated for a channel during a call. Resolves: #35 ASTERISK-29992 UserNote: A "dialmode" option has been added which allows specifying, on a per-channel basis, what methods of subscriber dialing (pulse and/or tone) are permitted. Additionally, this can be changed on a channel at any point during a call using the CHANNEL function. (cherry picked from commit `82d7bb49dd`)	3 years ago
Joshua Colp	9c5f5358d7	Revert "app_queue: periodic announcement configurable start time." This reverts commit `3fd0b65bae`. Reason for revert: Causes segmentation fault. Change-Id: Ic189c6f7872943a5500d3e71142f0c09d54fcc31 (cherry picked from commit `de15852ef0`)	3 years ago
Naveen Albert	99d17db963	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 `b33f92cbb5`)	3 years ago
Naveen Albert	c9aa3b0169	voicemail.conf: Fix incorrect comment about #include. A comment at the top of voicemail.conf says that #include cannot be used in voicemail.conf because this breaks the ability for app_voicemail to auto-update passwords. This is factually incorrect, since Asterisk has no problem updating files that are #include'd in the main configuration file, and this does work in voicemail.conf as well. ASTERISK-30479 #close Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3 (cherry picked from commit `d0f39250ee`)	3 years ago
Jaco Kroon	d6e733d4bc	app_queue: periodic announcement configurable start time. This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. ASTERISK-30437 #close Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0 Signed-off-by: Jaco Kroon <jaco@uls.co.za> (cherry picked from commit `3fd0b65bae`)	3 years ago
Holger Hans Peter Freyther	723c9093ad	res_http_media_cache: Introduce options and customize Make the existing CURL parameters configurable and allow to specify the usable protocols, proxy and DNS timeout. ASTERISK-30340 Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b (cherry picked from commit `8f088aa0f7`)	3 years ago
Sean Bright	aef0c0ce0e	app_queue: Reset all queue defaults before reload. Several queue fields were not being set to their default value during a reload. Additionally added some sample configuration options that were missing from queues.conf.sample. Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4	3 years ago
Naveen Albert	a1da8042d1	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
Michael Kuron	fee9012fe1	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	b9c031c1f8	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	3 years ago
Naveen Albert	531eacd6c9	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	3 years ago
Mike Bradeen	81f10e847e	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
Naveen Albert	5ede4e217a	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	3 years ago
Henning Westerholt	12445040d3	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
Naveen Albert	b331caca30	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
George Joseph	2a500b325a	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	3 years ago
Naveen Albert	8c791f9a65	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	3 years ago
sungtae kim	80bc844fd6	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	3 years ago
Ben Ford	881a3f2306	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
Naveen Albert	205c7c8d21	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	3 years ago
George Joseph	c799db6a21	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	4ffc5561c4	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
George Joseph	2d5a6498dd	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	3 years ago
Joshua C. Colp	a0713a9f70	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
Naveen Albert	3fa66c92b5	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
George Joseph	8a8416e365	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	3 years ago
Naveen Albert	5feebc0857	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	3 years ago
Sam Banks	be6a03f68c	queues.conf.sample: Correction of typo ASTERISK-30126 #close Change-Id: I009c4dcbf9338a13e3baf87b52a5bbe4f9f81a42	3 years ago
George Joseph	1fa568e76f	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	3 years ago
George Joseph	639d72e98c	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	3 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Kevin Harwell	a3b2daf127	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	3 years ago
Naveen Albert	31dc28ab09	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	3 years ago
Michael Cargile	a2679b0ee2	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	4 years ago
Naveen Albert	19c841950b	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	4 years ago
Naveen Albert	bce722e60d	samples: Remove obsolete sample configs. Removes a couple sample config files for modules which have since been removed from Asterisk. ASTERISK-30008 #close Change-Id: I6be89cafc6c575d98a5315e4912b61a833aacf52	4 years ago
Mark Petersen	1cdaeb8161	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	4 years ago
Kevin Harwell	272bac70dd	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	4 years ago
Joshua C. Colp	fdc1c750f3	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	4 years ago
Ben Ford	0724b767a3	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	4 years ago
Joshua C. Colp	4aedaaadeb	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	4 years ago
Naveen Albert	ede4e2099f	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	4 years ago
Naveen Albert	94df607771	build: Remove obsolete leftover build references. Removes some leftover build and config references to modules that have since been removed from Asterisk. ASTERISK-29935 #close Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af	4 years ago
Naveen Albert	2ba5da15b0	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	4 years ago
Alexei Gradinari	c12cb899de	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	4 years ago
Naveen Albert	585c2d17bb	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	4 years ago

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2120 Commits (244e0b69ea776d4be35de77b99aa9329a1a35dbd)