772 Commits (531eb65cf307c1852d7e27da2b2b8daedde488e9)

Author	SHA1	Message	Date
Ben Ford	55c53de022	logging: Add .log to samples and update asterisk.logrotate. ... Added .log extension to the sample logs in logger.conf.sample so that they will be able to be opened in the browser when attached to JIRA tickets. Because of this, asterisk.logrotate has also been updated to look for .log extensions instead of no extension for log files such as full and messages. Change-Id: I5de743c03f08047d6c6cc80cac5019ae0c4c200f	5 years ago
Mark Murawski	b4347c4861	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	71dfbdc7b9	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	fc03116d9b	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	3330fb41f4	xml: Allow deprecated_in and removed_in for MODULEINFO. ... ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	5 years ago
Joshua C. Colp	149e5e5b86	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
Jaco Kroon	b0f349a330	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	6e695c867f	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	5 years ago
Alexander Traud	389b8b0774	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	91b0778791	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	55891227e8	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	6 years ago
lvl	b08427134f	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, and changes the default location from /tmp to /var/cache/asterisk. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	6 years ago
Joshua C. Colp	eda3679c1c	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	6 years ago
Sean Bright	d04b5903d1	CHANGES: Remove already applied CHANGES update ... Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	6 years ago
Kevin Harwell	56028426de	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	6 years ago
George Joseph	773f424c7f	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	6 years ago
Sean Bright	52ca2323aa	chan_sip.c: Don't build by default ... ASTERISK-29083 #close Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003	6 years ago
Kevin Harwell	3c4a1722b6	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	6 years ago
George Joseph	54ddf19141	logger.c: Added a new log formatter called "plain" ... Added a new log formatter called "plain" that always prints file, function and line number if available (even for verbose messages) and never prints color control characters. It also doesn't apply any special formatting for verbose messages. Most suitable for file output but can be used for other channels as well. You use it in logger.conf like so: debug => [plain]debug console => [plain]error,warning,debug,notice,pjsip_history messages => [plain]warning,error,verbose Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d	6 years ago
Asterisk Development Team	1f5e6805bf	Update CHANGES and UPGRADE.txt for 18.0.0	6 years ago
Ben Ford	5fbed5af24	res_stir_shaken: Add stir_shaken option and general improvements. ... Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	6 years ago
Walter Doekes	312c23b0e1	app_queue: (Breaking change) shared_lastcall and autofill default to no ... If your queues.conf had _no_ [general] section, they would default to 'yes'. Now, they always default to 'no'. (Actually, commit ed615afb7e already partially fixed it for shared_lastcall.) ASTERISK-28951 Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6	6 years ago
sungtae kim	81b5e4a73f	res_pjsip.c: Added disable_rport option for pjsip.conf ... Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	6 years ago
Kevin Harwell	cfed0ea033	manager - Add Content-Type parameter to the SendText action ... This patch allows a user of AMI to now specify the type of message content contained within by setting the 'Content-Type' parameter. Note, the AMI version has been bumped for this change. ASTERISK-28945 #close Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb	6 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation ... The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	6 years ago
Joshua C. Colp	a143c3a7b7	res_sorcery_memory_cache: Disallow per-object expire with full backend. ... The AMI action and CLI command did not take into account the properties of full backend caching. This resulted in an expired object remaining removed until a full backend update occurred, instead of having the object updated when needed. This change makes it so that the AMI action and CLI command for object expire will now fail instead of putting the cache into an undesired state. If full backend caching is enabled then only operations which act on the entire cache are available. ASTERISK-28942 Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4	6 years ago
Joshua C. Colp	a7aaee70c6	res_pjsip_logger: Expand functionality to improve logging. ... The PJSIP packet logger now has the following CLI commands: pjsip set logger pcap <filename> When used this will create a pcap file containing the incoming and outgoing SIP packets, in unencrypted form. pjsip set logger verbose <on / off> This allows you to toggle logging to verbose on and off. pjsip set logger host <IP/subnet mask> add This allows you to add an additional IP address or subnet mask to logging, allowing you to log multiple instead of just a single IP address or all traffic. The normal "pjsip set logger host" CLI command has also been expanded to allow subnet masks as well. ASTERISK-28895 Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e	6 years ago
Joshua C. Colp	15cbff9d54	ari: Allow variables to be set on channel create. ... This change adds the same variable functionality that is available for originating a channel to the create call. Now when creating a channel you can specify dialplan variables to set instead of having to do another API call. ASTERISK-28896 Change-Id: If13997ba818136d7c070585504fc4164378aa992	6 years ago
Ben Ford	9acf840f7c	res_stir_shaken: Implemented signature verification. ... There are a lot of moving parts in this patch, but the focus of it is on the verification of the signature using a public key located at the public key URL provided in the JSON payload. First, we check the database to see if we have already downloaded the key. If so, check to see if it has expired. If it has, redownload from the URL. If we don't have an entry in the database, just go ahead and download the public key. The expiration is tested each time we download the file. After that, read the public key from the file and use it to verify the signature. All sanity checking is done when the payload is first received, so the verification is complete once this point is reached. The XML has also been added since a new config option was added to general (curl_timeout). The maximum amount of time to wait for a download can be configured through this option, with a low value by default. Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c	6 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Joshua C. Colp	6cfc6ff53c	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	6 years ago
Jean Aunis	de66713fd5	func_volume: Accept decimal number as argument ... Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	6 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Joshua C. Colp	3ed80fc57b	CHANGES: Change md file extension to txt. ... Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7 (cherry picked from commit c9cd681261)	6 years ago
Joshua C. Colp	21e9051461	res_pjsip_session: Apply intention behind requested formats. ... When an outgoing channel is created a list of formats may optionally be provided which is used as a request that the formats be used if possible. If an endpoint is not configured for any of the formats we ignore this request and use what is configured. This has the side effect of also including other stream types (such as video) that were not present in the requested formats. This change makes it so that the intention of the request is preserved - that is if only an audio format is requested then even if there is no joint audio format between the request and the configuration we will still only place an audio stream in the outgoing call. ASTERISK-28787 Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f	6 years ago
Jaco Kroon	82c3939c38	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. ... A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
George Joseph	99efe1f868	Merge "codec negotiation: add incoming_call_offer_prefs option"	6 years ago
Joshua Colp	e8468eee13	Merge "res_rtp_asterisk: Add 'rtp show settings' cli command"	6 years ago
Rodrigo Ramírez Norambuena	e089779908	res_rtp_asterisk: Add 'rtp show settings' cli command ... This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	6 years ago
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option ... Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Kevin Harwell	a715cf5aaa	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Kevin Harwell	999fdef335	Merge "app_mixmonitor: Turn on synchronization by default"	6 years ago
Sean Bright	8dcdce42a9	app_mixmonitor: Turn on synchronization by default ... The optional synchronization behavior created in 64906c4c9b is now the default for MixMonitor. * Add a new flag 'n' that allows for this behavior to be turned off * Add a notice when the 'S' option is used indicating that it is no longer necessary Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4	6 years ago
Sean Bright	ddfb60ac2c	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	6 years ago
George Joseph	a72caa041f	doc: Fix CHANGES entries to have .txt suffix and update READMEs ... Although the wiki page for the new CHANGES and UPGRADE scheme states that the files must have the ".txt" suffix, the READMEs didn't. Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05 (cherry picked from commit 7416703f04)	7 years ago
George Joseph	b76ab5e5c9	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	7 years ago
George Joseph	9688381f05	Merge "http: Add ability to disable /httpstatus URI"	7 years ago
Friendly Automation	95c6fbeae0	Merge "app_voicemail: Remove MessageExists and MESSAGE_EXISTS()"	7 years ago
Sean Bright	0dce6f746b	http: Add ability to disable /httpstatus URI ... Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	7 years ago
Sean Bright	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() ... * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit ec83b11183) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit fd64bb66f9) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	7 years ago