804 Commits (cb1dfecc1164b54bdddc94de1a5520866f561ed6)

Author	SHA1	Message	Date
Naveen Albert	cb1dfecc11	app_originate: Add ability to set codecs ... A list of codecs to use for dialplan-originated calls can now be specified in Originate, similar to the ability in call files and the manager action. Additionally, we now default to just using the slin codec for originated calls, rather than all the slin* codecs up through slin192, which has been known to cause issues and inconsistencies from AMI and call file behavior. ASTERISK-29543 Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883	4 years ago
Asterisk Development Team	288d018fb7	Update CHANGES and UPGRADE.txt for 18.6.0	4 years ago
Naveen Albert	118d848238	func_frame_drop: New function ... Adds function to selectively drop specified frames in the TX or RX direction on a channel, including control frames. ASTERISK-29478 Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec	4 years ago
Naveen Albert	940f6c4a03	app_queue: Allow streaming multiple announcement files ... Allows multiple files comprising an agent announcement to be played by separating on the ampersand, similar to the multi-file support in other Asterisk applications. ASTERISK-29528 Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a	5 years ago
Igor Goncharovsky	1e4ed61a2b	res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern ... PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request. It may be used to get all X- headers in case the actual set and names of headers unknown. ASTERISK-29389 Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b	5 years ago
Rijnhard Hessel	71dd1d91ad	res_statsd: handle non-standard meter type safely ... Meter types are not well supported, lacking support in telegraf, datadog and the official statsd servers. We deprecate meters and provide a compliant fallback for any existing usages. A flag has been introduced to allow meters to fallback to counters. ASTERISK-29513 Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7	5 years ago
Naveen Albert	016f6a0e14	app_dtmfstore: New application to store digits ... Adds application to asynchronously collect digits dialed on a channel in the TX or RX direction using a framehook and stores them in a specified variable, up to a configurable number of digits. ASTERISK-29477 Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f	5 years ago
Asterisk Development Team	993b3ba919	Update CHANGES and UPGRADE.txt for 18.5.1	5 years ago
Naveen Albert	a41d192e99	app_reload: New Reload application ... Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	5 years ago
Naveen Albert	1b21b1abf7	app_waitforcond: New application ... While several applications exist to wait for a certain event to occur, none allow waiting for any generic expression to become true. This application allows for waiting for a condition to become true, with configurable timeout and checking interval. ASTERISK-29444 Change-Id: I08adf2824b8bc63405778cf355963b5005612f41	5 years ago
Andre Barbosa	283812e492	res_stasis_playback: Send PlaybackFinish event only once for errors ... When we try to play a list of sound files in the same Play command, we get only one PlaybackFinish event, after all sounds are played. But in the case where the Play fails (because channel is destroyed for example), Asterisk will send one PlaybackFinish event for each sound file still to be played. If the list is big, Asterisk is sending many events. This patch adds a failed state so we can understand that the play failed. On that case we don't send the event, if we still have a list of sounds to be played. When we reach the last sound, we send the PlaybackFinish with the failed state. ASTERISK-29464 #close Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322	5 years ago
Naveen Albert	c4236dcff2	app_dial: Expanded A option to add caller announcement ... Hitherto, the A option has made it possible to play audio upon answer to the called party only. This option is expanded to allow for playback of an audio file to the caller instead of or in addition to the audio played to the answerer. ASTERISK-29442 Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e	5 years ago
Asterisk Development Team	0747162d4f	Update CHANGES and UPGRADE.txt for 18.5.0	5 years ago
Naveen Albert	a611a0cd42	app_originate: Allow setting Caller ID and variables ... Caller ID can now be set on the called channel and Variables can now be set on the destination using the Originate application, just as they can be currently using call files or the Manager Action. ASTERISK-29450 Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66	5 years ago
Naveen Albert	a40e58a4da	app_confbridge: New ConfKick() application ... Adds a new ConfKick() application, which may be used to kick a specific channel, all channels, or all non-admin channels from a specified conference bridge, similar to existing CLI and AMI commands. ASTERISK-29446 Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b	5 years ago
Naveen Albert	99573f9540	res_pjsip_dtmf_info: Hook flash ... Adds hook flash recognition support for application/hook-flash. ASTERISK-29460 Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea	5 years ago
Naveen Albert	a861522467	app_confbridge: New option to prevent answer supervision ... A new user option, answer_channel, adds the capability to prevent answering the channel if it hasn't already been answered yet. ASTERISK-29440 Change-Id: I26642729d0345f178c7b8045506605c8402de54b	5 years ago
George Joseph	8e2672d2a4	res_pjsip_messaging: Refactor outgoing URI processing ... * Implemented the new "to" parameter of the MessageSend() dialplan application. This allows a user to specify a complete SIP "To" header separate from the Request URI. * Completely refactored the get_outbound_endpoint() function to actually handle all the destination combinations that we advertized as supporting. * We now also accept a destination in the same format as Dial()... PJSIP/number@endpoint * Added lots of debugging. ASTERISK-29404 Reported by Brian J. Murrell Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce	5 years ago
Naveen Albert	9106c9d1f1	func_math: Three new dialplan functions ... Introduces three new dialplan functions, MIN and MAX, which can be used to calculate the minimum or maximum of up to two numbers, and ABS, an absolute value function. ASTERISK-29431 Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d	5 years ago
Naveen Albert	033c2a2283	func_volume: Add read capability to function. ... Up until now, the VOLUME function has been write only, so that TX/RX values can be set but not read afterwards. Now, previously set TX/RX values can be read later. ASTERISK-29439 Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f	5 years ago
Joseph Nadiv	b21d4d1b87	res_pjsip.c: Support endpoints with domain info in username ... In multidomain environments, it is desirable to create PJSIP endpoints with the domain info in the endpoint name in pjsip_endpoint.conf. This resulted in an error with registrations, NOTIFY, and OPTIONS packet generation. This commit will detect if there is an @ in the endpoint identifier and generate the URI accordingly so NOTIFY and OPTIONS From headers will generate correctly. ASTERISK-28393 Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619	5 years ago
Jeremy Lainé	0f8e2174a7	res_rtp_asterisk: make it possible to remove SOFTWARE attribute ... By default Asterisk reports the PJSIP version in a SOFTWARE attribute of every STUN packet it sends. This may not be desired in a production environment, and RFC5389 recommends making the use of the SOFTWARE attribute a configurable option: https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2 This patch adds a `stun_software_attribute` yes/no option to make it possible to omit the SOFTWARE attribute from STUN packets. ASTERISK-29434 Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b	5 years ago
Joseph Nadiv	83c2a16b2e	res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml ... RFC 4235 Section 4.1.6 describes XML elements that should be sent to subscribed endpoints to identify the local and remote participants in the dialog. This patch adds this functionality to PJSIP by iterating through the ringing channels causing the NOTIFY, and inserts the channel info into the dialog so that information is properly passed to the endpoint in dialog-info+xml. ASTERISK-24601 Patch submitted: Joshua Elson Modified by: Joseph Nadiv and Sean Bright Tested by: Joseph Nadiv Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b	5 years ago
Naveen Albert	bfc25e5de2	app_voicemail: Configurable voicemail beep ... Hitherto, VoiceMail() played a non-customizable beep tone to indicate the caller could leave a message. In some cases, the beep may not be desired, or a different tone may be desired. To increase flexibility, a new option allows customization of the tone. If the t option is specified, the default beep will be overridden. Supplying an argument will cause it to use the specified file for the tone, and omitting it will cause it to skip the beep altogether. If the option is not used, the default behavior persists. ASTERISK-29349 Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280	5 years ago
Naveen Albert	0ad3504ce0	AMI: Add AMI event to expose hook flash events ... Although Asterisk can receive and propogate flash events, it currently provides no mechanism for doing anything with them itself. This AMI event allows flash events to be processed by Asterisk. Additionally, AST_CONTROL_FLASH is included in a switch statement in channel.c to avoid throwing a warning when we shouldn't. ASTERISK-29380 Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81	5 years ago
Ben Ford	a84d34035a	STIR/SHAKEN: Switch to base64 URL encoding. ... STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	5 years ago
Ben Ford	5e6508b56f	STIR/SHAKEN: Fix certificate type and storage. ... During OpenSIPit, we found out that the public certificates must be of type X.509. When reading in public keys, we use the corresponding X.509 functions now. We also discovered that we needed a better naming scheme for the certificates since certificates with the same name would cause issues (overwriting certs, etc.). Now when we download a public certificate, we get the serial number from it and use that as the name of the cached certificate. The configuration option public_key_url in stir_shaken.conf has also been renamed to public_cert_url, which better describes what the option is for. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d	5 years ago
George Joseph	40bdfff73b	Updates for the MessageSend Dialplan App ... Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	5 years ago
Asterisk Development Team	1949d828b7	Update CHANGES and UPGRADE.txt for 18.4.0	5 years ago
Sean Bright	d2dcd15bd8	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	5 years ago
Mark Murawski	a0009c807e	logger: Console sessions will now respect logger.conf dateformat= option ... The 'core' console (ie: asterisk -c) does read logger.conf and does use the dateformat= option. Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf and uses a hard coded dateformat option for printing received verbose messages: main/logger.c: static char dateformat[256] = "%b %e %T" This change will load logger.conf for each remote console session and use the dateformat= option to set the per-line timestamp for verbose messages Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1 ASTERISK-25358: #close Reported-by: Igor Liferenko	5 years ago
Joshua C. Colp	f213833514	res_pjsip: Add support for partial transport reload. ... Some configuration items for a transport do not result in the underlying transport changing, but instead are just state we keep ourselves and use. It is perfectly reasonable to change these items. These include local_net and external_* information. ASTERISK-29354 Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a	5 years ago
Jaco Kroon	f47c5cbdf9	menuselect: exit non-zero in case of failure on --enable|disable options. ... ASTERISK-29348 Change-Id: I77e3466435f5a51a57538b29addb68d811af238d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Joshua C. Colp	60fb559ccc	xml: Allow deprecated_in and removed_in for MODULEINFO. ... ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	5 years ago
Joshua C. Colp	60800b038a	xml: Embed module information into core XML documentation. ... This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	5 years ago
Asterisk Development Team	0afd37e3b5	Update CHANGES and UPGRADE.txt for 18.3.0	5 years ago
Jaco Kroon	bee35fe04a	func_odbc: Introduce minargs config and expose ARGC in addition to ARGn. ... minargs enables enforcing of minimum count of arguments to pass to func_odbc, so if you're unconditionally using ARG1 through ARG4 then this should be set to 4. func_odbc will generate an error in this case, so for example [FOO] minargs = 4 and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a potentially leaked ARG4 from Gosub(). ARGC is needed if you're using optional argument, to verify whether or not an argument has been passed, else it's possible to use a leaked ARGn from Gosub (app_stack). So now you can safely do ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Sebastien Duthil	092628c982	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	5 years ago
Alexander Traud	703158b903	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
George Joseph	28f187d6c5	chan_iax2.c: Require secret and auth method if encryption is enabled ... If there's no secret specified for an iax2 peer and there's no secret specified in the dial string, Asterisk will crash if the auth method requested by the peer is MD5 or plaintext. You also couldn't specify a default auth method in the [general] section of iax.conf so if you don't have static peers defined and just use the dial string, Asterisk will still crash even if you have a secret specified in the dial string. * Added logic to iax2_call() and authenticate_reply() to print a warning and hanhup the call if encryption is requested and there's no secret or auth method. This prevents the crash. * Added the ability to specify a default "auth" in the [general] section of iax.conf. ASTERISK-29624 Reported by: N A Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254	5 years ago
Dan Cropp	088816284a	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	5 years ago
Asterisk Development Team	89fea9bafe	Update CHANGES and UPGRADE.txt for 18.2.0	5 years ago
lvl	92fcd4edba	Introduce astcachedir, to be used for temporary bucket files ... As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, but leaves it at /tmp by default, to ensure backwards compatibility. A future commit that only targets master could change the default location to something more sensible such as /var/tmp/asterisk. At that point, the cachedir could be created and cleaned up during uninstall by the Makefile script. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	5 years ago
Joshua C. Colp	fd57fae048	voicemail: add option 'e' to play greetings as early media ... When using this option, answering the channel is deferred until all prompts/greetings have been played and the caller is about to leave their message. ASTERISK-29118 #close Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb	5 years ago
Sean Bright	994fbdaf48	CHANGES: Remove already applied CHANGES update ... Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	5 years ago
Asterisk Development Team	98d1537c1e	Update CHANGES and UPGRADE.txt for 18.1.0	5 years ago
Kevin Harwell	6255e7976c	Logging: Add debug logging categories ... Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849 (cherry picked from commit 56028426de)	5 years ago
George Joseph	4a049ad510	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
Asterisk Development Team	f589985840	Update CHANGES and UPGRADE.txt for 18.0.0	5 years ago
Kevin Harwell	31fbfc5e95	chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution ... This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function is called on a channel prior to answering a warning is issued and the function returns unsuccessful. ASTERISK-28878 #close Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb	5 years ago